[asterisk-users] Call files without permission for asterisk to read
Hi all, I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this problem? I need to continue the execution of call files on secondary server if primary server fails. The call files are suppose to retry for 45 mins if the call does not get connected. Thanks in advance. -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- 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] Call files without permission for asterisk to read
Thanks for the responses. Touching a file after setting permissions does not work. Asterisk only looks at the new file only, not all the files in the directory. Restarting asterisk does work, but dont want to do this. Best way i think would be, as suggested by JG, to sync in a tmp directory and at the time of switch-over mv to outgoing directory. Cheers On Fri, Nov 22, 2013 at 12:36 AM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 21 Nov 2013, Rizwan Hisham wrote: Hi all,I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this problem? When you activate the secondary, 'touch' the files in the spool directory. -- 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 -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- 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] Realtime Call Files
Hi all, Is there any way of originating calls in future without using call files? We have 2 servers (1 active at a time). If we use call files with modification date in future, on the 1st server and it dies and, we activate the second server but we lose the call files. I could have a cronjob on both servers and create callfiles reading execution time from database, but this involves some other complications. Any crazy ideas would be helpful. Thanks -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- 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] Auto Redial Unconditional
Hi All, I need a softphone (PC/Mobile) which does auto redial in any case (noanswer, answer, busy, congestion etc) after a given time interval. So if the time interval was 5 secs, it would dial last number dialled after every hangup (or every failure to dial). Does anyone know such feature in a softphone? -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- 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] Call Transfer not working
Thanks everyone. I was using the Tt flag but in the wrong place in the dial application. Cheers On Mon, Apr 9, 2012 at 4:54 PM, Takehiro Matsushima takehiro.dream...@gmail.com wrote: Thank you so much. OK, I understood that to transfer the call t is usually used, is it right? And I should write so in my last mail. t and T are described with same sentences in official wiki... Regards, Takehiro Matsushima 2012/4/9 Chris Bagnall aster...@lists.minotaur.cc: On 9/4/12 3:04 am, Takehiro Matsushima wrote: // I don't know what's difference t and T. T allows the caller to transfer. t allows the called user to transfer. You very rarely want Tt - since I doubt you want an incoming caller to be able to transfer their call all over the place. You usually want t on incoming calls and T on outgoing calls. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Background music during a call
Very nice Loan. Here is the chat-room dialplan with a little tweek which
lets you set the volume up/down or mute/unmute the song.
Use 4/6 to increase/decrease the volume and 2/8 to mute/unmute the song
[chat-room]
exten = love,1,Goto(love-a,1)
exten = love,2,Goto(love-b,1)
exten = love-a,1,Set(__MOH=love)
exten = love-a,n,Dial(Local/fake@chat-
room,,G(chat-room,chat,1))
exten = love-b,1,Goto(chat,100)
exten = curse,1,Goto(curse-a,1)
exten = curse,2,Goto(curse-b,1)
exten = curse-a,1,Set(__MOH=curse)
exten = curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
exten = curse-b,1,Goto(chat,100)
exten = fake,1,Answer
exten = fake,2,MusicOnHold(${MOH})
exten = chat,1,Goto(100)
exten = chat,2,MeetMe(${MM},dx1q)
exten = chat,100,MeetMe(${MM},daAx1q)
exten = h,1,MeetMeAdmin(${MM},K)
exten = 4,1,MeetMeAdmin(${MM},t,2)
exten = 6,1,MeetMeAdmin(${MM},T,2)
exten = 2,1,MeetMeAdmin(${MM},M,2)
exten = 8,1,MeetMeAdmin(${MM},m,2)
exten= _X,2,Goto(chat-room,chat,100)
Here channel 2 always seem to be the one playing the MOH, thats why its hard
coded into the MeetMeAdmin application.
If there is a another way to know which channel is playing the song then
please do let me know.
Cheers
On Tue, May 10, 2011 at 9:33 AM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Thanks a lot loan. Will try it today.
Cheers
On Mon, May 9, 2011 at 6:25 PM, Ioan Indreias indre...@gmail.com wrote:
Updated dialplan: fix a typo when using MOH variable and now you have
truly dynamic conference rooms.
Have fun,
Ioan.
+
exten = _[12]XXX,1,Set(__MM=${EPOCH})
exten = _1XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,love,1))
exten = _2XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,curse,1))
[chat-room]
exten = love,1,Goto(love-a,1)
exten = love,2,Goto(love-b,1)
exten = love-a,1,Set(__MOH=love)
exten = love-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
exten = love-b,1,Goto(chat,100)
exten = curse,1,Goto(curse-a,1)
exten = curse,2,Goto(curse-b,1)
exten = curse-a,1,Set(__MOH=curse)
exten = curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
exten = curse-b,1,Goto(chat,100)
exten = fake,1,Answer
exten = fake,2,MusicOnHold(${MOH})
exten = chat,1,Goto(100)
exten = chat,2,MeetMe(${MM},dx1q)
exten = chat,100,MeetMe(${MM},daAx1q)
exten = h,1,MeetMeAdmin(${MM},K)
+
On Mon, May 9, 2011 at 4:02 PM, Ioan Indreias indre...@gmail.com wrote:
I have tested the following dialplan and it could be used as a
starting point. What you have to resolve is how to generate different
MeetMe conference room - in the example we have only one room = 1234
If you prefix the dialled extension with 1 = you will have a lovely
chat. With 2 - cursing chat.
HTH,
Ioan
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Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
--
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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Re: [asterisk-users] Background music during a call
Ooops,
here is the correct version, Missed the capital X option in meetme before
which lets you control the volume etc.
[chat-room]
exten = love,1,Goto(love-a,1)
exten = love,2,Goto(love-b,1)
exten = love-a,1,Set(__MOH=love)
exten = love-a,n,Dial(Local/fake@chat-
room,,G(chat-room,chat,1))
exten = love-b,1,Goto(chat,100)
exten = curse,1,Goto(curse-a,1)
exten = curse,2,Goto(curse-b,1)
exten = curse-a,1,Set(__MOH=curse)
exten = curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
exten = curse-b,1,Goto(chat,100)
exten = fake,1,Answer
exten = fake,2,MusicOnHold(${MOH})
exten = chat,1,Goto(100)
exten = chat,2,MeetMe(${MM},dx1qX)
exten = chat,100,MeetMe(${MM},daAx1qX)
exten = h,1,MeetMeAdmin(${MM},K)
exten = 4,1,MeetMeAdmin(${MM},t,2)
exten = 6,1,MeetMeAdmin(${MM},T,2)
exten = 2,1,MeetMeAdmin(${MM},M,2)
exten = 8,1,MeetMeAdmin(${MM},m,2)
exten= _X,2,Goto(chat-room,chat,100)
On Tue, May 10, 2011 at 9:57 PM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Very nice Loan. Here is the chat-room dialplan with a little tweek which
lets you set the volume up/down or mute/unmute the song.
Use 4/6 to increase/decrease the volume and 2/8 to mute/unmute the song
[chat-room]
exten = love,1,Goto(love-a,1)
exten = love,2,Goto(love-b,1)
exten = love-a,1,Set(__MOH=love)
exten = love-a,n,Dial(Local/fake@chat-
room,,G(chat-room,chat,1))
exten = love-b,1,Goto(chat,100)
exten = curse,1,Goto(curse-a,1)
exten = curse,2,Goto(curse-b,1)
exten = curse-a,1,Set(__MOH=curse)
exten = curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
exten = curse-b,1,Goto(chat,100)
exten = fake,1,Answer
exten = fake,2,MusicOnHold(${MOH})
exten = chat,1,Goto(100)
exten = chat,2,MeetMe(${MM},dx1q)
exten = chat,100,MeetMe(${MM},daAx1q)
exten = h,1,MeetMeAdmin(${MM},K)
exten = 4,1,MeetMeAdmin(${MM},t,2)
exten = 6,1,MeetMeAdmin(${MM},T,2)
exten = 2,1,MeetMeAdmin(${MM},M,2)
exten = 8,1,MeetMeAdmin(${MM},m,2)
exten= _X,2,Goto(chat-room,chat,100)
Here channel 2 always seem to be the one playing the MOH, thats why its
hard coded into the MeetMeAdmin application.
If there is a another way to know which channel is playing the song then
please do let me know.
Cheers
On Tue, May 10, 2011 at 9:33 AM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Thanks a lot loan. Will try it today.
Cheers
On Mon, May 9, 2011 at 6:25 PM, Ioan Indreias indre...@gmail.com wrote:
Updated dialplan: fix a typo when using MOH variable and now you have
truly dynamic conference rooms.
Have fun,
Ioan.
+
exten = _[12]XXX,1,Set(__MM=${EPOCH})
exten = _1XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,love,1))
exten = _2XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,curse,1))
[chat-room]
exten = love,1,Goto(love-a,1)
exten = love,2,Goto(love-b,1)
exten = love-a,1,Set(__MOH=love)
exten = love-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
exten = love-b,1,Goto(chat,100)
exten = curse,1,Goto(curse-a,1)
exten = curse,2,Goto(curse-b,1)
exten = curse-a,1,Set(__MOH=curse)
exten = curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
exten = curse-b,1,Goto(chat,100)
exten = fake,1,Answer
exten = fake,2,MusicOnHold(${MOH})
exten = chat,1,Goto(100)
exten = chat,2,MeetMe(${MM},dx1q)
exten = chat,100,MeetMe(${MM},daAx1q)
exten = h,1,MeetMeAdmin(${MM},K)
+
On Mon, May 9, 2011 at 4:02 PM, Ioan Indreias indre...@gmail.com
wrote:
I have tested the following dialplan and it could be used as a
starting point. What you have to resolve is how to generate different
MeetMe conference room - in the example we have only one room = 1234
If you prefix the dialled extension with 1 = you will have a lovely
chat. With 2 - cursing chat.
HTH,
Ioan
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Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
--
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
--
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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Re: [asterisk-users] Background music during a call
Its in testing phase right now Ioan. Also the curse thing is just an idea. We may not implement it actually, or maybe we do in future but not now. Lets see. On Tue, May 10, 2011 at 10:23 PM, Ioan Indreias indre...@gmail.com wrote: Glad to know it works for you. I would like to hear your love/curse MOH - do you have some links to your mp3 files? :) BR, Ioan (with capital i) On Tue, May 10, 2011 at 6:59 PM, Rizwan Hisham rizwanhas...@gmail.com wrote: Ooops, here is the correct version, Missed the capital X option in meetme before which lets you control the volume etc. -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Background music during a call
Thanks for the reply. I looked into the G option of Dial applications. No problem with that but How do I create a ghost call? My dial plan will look like this: Caller A calls Caller B normally: exten= _XXX,1,SomePreDialApps() exten= _XXX,n,Dial(SIP/B) exten= _XXX,n,Hangup() Caller A calls caller B ith background music exten= _*9XXX,1,SomePreDialApps() exten= _*9XXX,n,Dial(SIP/B,,G(10)) exten= _*9XXX,n,Hangup() exten= _*9XXX,10,Goto(mm,1,1) exten= _*9XXX,11,Goto(mm,1,1) Waiting for your replies Thanks On Fri, May 6, 2011 at 10:37 PM, Ioan Indreias indre...@gmail.com wrote: On Fri, May 6, 2011 at 6:30 PM, Rizwan Hisham rizwanhas...@gmail.com wrote: I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). Let's start with your actual dialplan (without the background music) and we could start from that point. Hint: I am planning to use option G of the Dial application + a meetme room where a ghost call will play the specified MOH class (lovely/cursing). HTH, Ioan. -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Background music during a call
Will this work: exten= 123,1,Meetme(1234) exten= 123,n,Hangup() exten= 5000,1,Dial(Local/123@bk_music/n,,m()) exten= 5000,2,Goto(bk_music,123,1) Parties can call 123 to enter a meeting room. and with the help of a callfile ic an dial a local channel to 5000 extension which in return calls a local channel to exten 123 to enter meet me. The dial command with second local channel will use m() option with moh call defind for each caller. will ring indefinately with moh and conf members will listen to it. Not tested it yet. Just sharing, will try it and let you know list. Cheers Mon, May 9, 2011 at 8:47 AM, Rizwan Hisham rizwanhas...@gmail.com wrote: Thanks for the reply. I looked into the G option of Dial applications. No problem with that but How do I create a ghost call? My dial plan will look like this: Caller A calls Caller B normally: exten= _XXX,1,SomePreDialApps() exten= _XXX,n,Dial(SIP/B) exten= _XXX,n,Hangup() Caller A calls caller B ith background music exten= _*9XXX,1,SomePreDialApps() exten= _*9XXX,n,Dial(SIP/B,,G(10)) exten= _*9XXX,n,Hangup() exten= _*9XXX,10,Goto(mm,1,1) exten= _*9XXX,11,Goto(mm,1,1) Waiting for your replies Thanks On Fri, May 6, 2011 at 10:37 PM, Ioan Indreias indre...@gmail.com wrote: On Fri, May 6, 2011 at 6:30 PM, Rizwan Hisham rizwanhas...@gmail.com wrote: I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). Let's start with your actual dialplan (without the background music) and we could start from that point. Hint: I am planning to use option G of the Dial application + a meetme room where a ghost call will play the specified MOH class (lovely/cursing). HTH, Ioan. -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Background music during a call
Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the two parties will enjoy/maybe not. I could'nt make it to work. Anyways, I think this feature will require some creative dialplanning as its not supported by default by the software. If anyone can tell me how to create a ghost call then the rest I may be able to figure out myself. If there is another way plz share coz im on a deadline. I am a wee bit of a programmer also, so if your idea needs changes in the code please dont hesitate to share, otherwise you WILL get a call from me with a special background noise crafted just for you :) Meanwhile i'll try my best to come up with a solution. Cheers -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] odbc error - server is gone
isql? On Sat, Apr 30, 2011 at 6:18 PM, Pezhman Lali l...@lopl.net wrote: check your odbc connection with isql best On Fri, Apr 29, 2011 at 9:33 PM, Warren Selby wcse...@selbytech.comwrote: You're using 1.4.2. Why not try upgrading to a more recent release of 1.4 (I believe 1.4.41 is current) and see if your issue has been resolved. Thanks, --Warren Selby, dCAP On Apr 29, 2011, at 7:32 AM, Rizwan Hisham rizwanhas...@gmail.com wrote: Yes I have it there, here the content of the file: i think the code is buggy, here is a comment from the function which generated the error (ast_odbc_smart_execute in res_odbc.c line 155 ) /* This is a really bad method of trying to correct a dead connection. It * only ever really worked with MySQL. It will not work with any other * database, since most databases prepare their statements on the server, * and if you disconnect, you invalidate the statement handle. Hence, if * you disconnect, you're going to fail anyway, whether you try to execute * a second time or not. */ This function is used all over. On Fri, Apr 29, 2011 at 12:23 AM, Sherwood McGowan sherwood.mcgo...@gmail.com sherwood.mcgo...@gmail.com wrote: On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham rizwanhas...@gmail.com rizwanhas...@gmail.com wrote: Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemailand herehttp://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage . I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for mysql on the server. I successfully completed the conversion of a lot of voicemail users into db yesterday. But today on the CLI thsi error was showing; [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL Execute error! [SELECT COUNT(*) FROM voicemessages WHERE dir = '/var/spool/asterisk/voicemail/default/1757XXX/INBOX'] I know that the error is caused due to stale odbc connection with mysql. But i want to find out if there is a cure for it. Why the connection went stale in the first place also. Any ideas? -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.comrizwanhas...@gmail.com W: http://www.axvoice.com/www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users do you have sanitysql = select 1 configured in res_odbc.ini? -- Sherwood McGowan Telecommunications and VOIP Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.comrizwanhas...@gmail.com W: http://www.axvoice.com/www.axvoice.com -- _ -- 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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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
Re: [asterisk-users] odbc error - server is gone
Yes I have it there, here the content of the file: i think the code is buggy, here is a comment from the function which generated the error (ast_odbc_smart_execute in res_odbc.c line 155 ) /* This is a really bad method of trying to correct a dead connection. It * only ever really worked with MySQL. It will not work with any other * database, since most databases prepare their statements on the server, * and if you disconnect, you invalidate the statement handle. Hence, if * you disconnect, you're going to fail anyway, whether you try to execute * a second time or not. */ This function is used all over. On Fri, Apr 29, 2011 at 12:23 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham rizwanhas...@gmail.comwrote: Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail and here http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage . I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for mysql on the server. I successfully completed the conversion of a lot of voicemail users into db yesterday. But today on the CLI thsi error was showing; [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL Execute error! [SELECT COUNT(*) FROM voicemessages WHERE dir = '/var/spool/asterisk/voicemail/default/1757XXX/INBOX'] I know that the error is caused due to stale odbc connection with mysql. But i want to find out if there is a cure for it. Why the connection went stale in the first place also. Any ideas? -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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 do you have sanitysql = select 1 configured in res_odbc.ini? -- Sherwood McGowan Telecommunications and VOIP Consultant -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] odbc error - server is gone
Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail and here http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage. I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for mysql on the server. I successfully completed the conversion of a lot of voicemail users into db yesterday. But today on the CLI thsi error was showing; [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL Execute error! [SELECT COUNT(*) FROM voicemessages WHERE dir = '/var/spool/asterisk/voicemail/default/1757XXX/INBOX'] I know that the error is caused due to stale odbc connection with mysql. But i want to find out if there is a cure for it. Why the connection went stale in the first place also. Any ideas? -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] CDR ARI Question
google for adaptive cdr. in asterisk.
On Sun, Apr 17, 2011 at 3:58 AM, John Jolly jgjo...@gmail.com wrote:
I have a particular DID that when called will prompt the user to enter the
caller id that they want to be displayed followed by it prompting for the
phone number to dial. How would I go about getting thest calls logged in
both CDR and ARI? Currently, only the callerid information from the original
caller is populated in CDR and ARI. Is there a way to get this call detail
to be logged in the CDR and ARI databases?
My current dialplan looks like this -- the calls do get recorded but there
is no link in ARI to the actual recorded file(s).
exten =
s,6,Set(CALLFILENAME=${calleridnum}--${STRFTIME(${EPOCH},,%d-%m-%Y-%H:$
exten = s,7,MixMonitor(${CALLFILENAME}.wav)
exten = s,8,Dial(SIP/flowroute/${telnum})
exten = s,9,StopMonitor()
Any help/advice is greatly appreciated...
J. Jolly
jgjolly[at]gmail.com
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Re: [asterisk-users] call-limit bypass
I am using asterisk 1.4.2 and it usually does enforce the limit. yesterday and couple of times before was an exception. I am still trying to find the reason behind. Any more suggestions please? oh by the way * 1.8.1.1 does enforce the call limit, i tested it yesteday on sip channels. On Mon, Apr 4, 2011 at 11:48 PM, Bryant Zimmerman brya...@zktech.comwrote: From what I understand on the newer versions of asterisk call-limit does not limit calls anymore. You have to limit them from your code using call groups. From what I have seen on the 1.6x and 1.8 versions call-limit does not limit your call counts. We use code and the GROUP_COUNT to limit calls. If you use it right it is rock solid. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- *From*: Rizwan Hisham rizwanhas...@gmail.com *Sent*: Monday, April 04, 2011 12:30 PM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: [asterisk-users] call-limit bypass Hi everyone, one of our users last night bypassed asterisk call-limit limitation. I have no Idea how. Is it possible? Is there a bug in asterisk that can be manipulated for this purpose? The call-limit variable was to 2, and the user initiated 169 calls in 2 minutes each has duration at least 8 minutes. Please comment... Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] call forwarding
Do this:
exten= _0522XX,1,Goto(${CONTEXT},0520${EXTEN:4:},1)
you can also use the dial command for this as well
exten= _0522XX,1,Dial(Local/0520${EXTEN:4}@${SOMECONTEXT})
replace ${CoNTEXT} and ${SOMECONTEXT} with name of your contexts which
contains 0520 numbers.
I have not tested it, you can try it on your setup.
On Mon, Apr 4, 2011 at 7:00 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hello list,
i have one question related to call forwarding.
i have 2 number for the inbound and i want to configure asterisk like that.
When the customer call the first number 0522XX the call will be
forwarding automatically to anther number 0520xx
Does anybody have a solution to this problem.
Thanks and Regards.
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Axvoice Inc.
V: +92 (0) 6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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[asterisk-users] call-limit bypass
Hi everyone, one of our users last night bypassed asterisk call-limit limitation. I have no Idea how. Is it possible? Is there a bug in asterisk that can be manipulated for this purpose? The call-limit variable was to 2, and the user initiated 169 calls in 2 minutes each has duration at least 8 minutes. Please comment... Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Fwd: Asterisk 1.6.2.10 CDR custom added field
You have to use adaptive cdr for this functionality. In 1.8 the conf file for adaptive cdr is cdr_adaptive_odbc.conf. The sample conf file should tell you everything. If you are using some other cdr engine then you will have to jump into the code of asterisk to make it log the item you want, which includes creating an extra variable in the cdr data struction, creating a function to set/get its value from dialplan, and then changing the sql command to include the extra variable for insertion into DB. On Thu, Mar 24, 2011 at 1:55 PM, Jonas Kellens jonas.kell...@telenet.bewrote: Hello, is there anyone who can point me to correct information ? Following http://pbxinaflash.com/forum/showthread.php?t=9042 and http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql Extending CDR does not result in a working environment for me. Any feedback appreciated. Kind regards, Jonas. Original Message Subject: [asterisk-users] Asterisk 1.6.2.10 CDR custom added field Date: Tue, 22 Mar 2011 14:05:23 +0100 From: Jonas Kellens jonas.kell...@telenet.be jonas.kell...@telenet.be Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com asterisk-users@lists.digium.com Hello list, I have added an extra field mycolumn to the cdr table in my MySQL-DB. I simply try to add a value to this column by doing the following in the dialplan : exten = 600,n,Set(CDR(mycolumn)=myvalue) But this value is not written to the column 'mycolumn' together with the other CDR-data. Why is this ?! Do I need further configuration ? (Not according to http://pbxinaflash.com/forum/showthread.php?t=9042) Kind regards, Jonas. -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Multiple Asterisk
Here is a better link for DUNDi http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/ skip the part which you know already On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes sf.ri...@gmail.comwrote: []'sf.rique On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger pabelan...@digium.comwrote: On 11-03-15 06:19 PM, Henrique Fernandes wrote: Have many diferenet locations that have convencional phones that need to call others locations with convencional phones. And we can not change this, I was reading and asterisk cannot handle it self this kind of setup, it needs an separated serrver to control and routers the calls to this poins right ? So can you guys give any help ? I guess asterisk with SER could do the job ? I don't believe SER will help you in the setup (see below). So my question is how do i make the 2 PABX with asterisk talk to each other? Do i need only 2 asterisk with digium or i need one server with SER to maki it happen ? There is another program that does what i am looking for ? If you require local hardware for each site, then you can install Asterisk at each location. You can then interconnect them using IAX2 or SIP, additionally you can use DUNDi in your dialplans to share information before the Asterisk boxes. Thanks! I had heard some thing about DUNDi but now i am reading i guess it is what i need! I am guessing i can use both IAX2 and SIP i read something about H.323 So i am gonna see which one is best to conect the Asterisk PBX if i am not able to use bot SIP and IAX2 Thanks! here is a link that explains better what DUNDi is! http://www.voip-info.org/wiki/view/DUNDi -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] AMI Timestamps unit
Never mind. Its in seconds :) On Tue, Mar 15, 2011 at 6:48 PM, Rizwan Hisham rizwanhas...@gmail.comwrote: Hi all, What is the unit of asterisk AMI events timestamp value? milli/micro etc ? -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] AMI Timestamps unit
Hi all, What is the unit of asterisk AMI events timestamp value? milli/micro etc ? -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] call being rejected
You can try changing the priority of '1104' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] tp this '1104' = 2. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] On Tue, Mar 15, 2011 at 6:59 PM, Jerry Geis ge...@pagestation.com wrote: I am using asterisk 1.8.3. I am getting this error: [Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected because extension not found in context 'smvoice-mediaport'. dialplan show gives me that the context is present: [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1104' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] 'mediaport_direct' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] 'public_address' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] the server is showing the call to 1104 which is valid : -- Executing [1104@smvoice-sip:1] Dial(SIP/528-0124, SIP/mndemo_to_vizioconfrm104/1104) in new stack == Using SIP RTP CoS mark 5 Why is my call not going through? Jerry -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] call being rejected
Isrlgb, Its the output from dialplan show command. The actual entry for the extension has to be like what you said. On Tue, Mar 15, 2011 at 7:16 PM, isr...@gmail.com wrote: Shouldn't that be Exten = 1104, 1, Goto(smvoice-mediaport-public-address,s,1) -Original Message- From: Rizwan Hisham rizwanhas...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 15 Mar 2011 19:03:33 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call being rejected -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] trunk not working if I register a phone at the same IP as the trunk peer's IP
1.8 supports static peers along with realtime peers. I have tested. On Mon, Feb 21, 2011 at 11:04 PM, Ricardo Carvalho rjcarvalho.li...@gmail.com wrote: Thanks Faisal, in fact I made a test that confirmed that in realtime asterisk doesn’t supported static peers, like you told me. Do you know if newer versions of asterisk, like 1.8, have this issue already solved? Regards, Ricardo. On Wed, Feb 16, 2011 at 6:26 PM, Faisal Hanif fai...@vopium.com wrote: I have played a lot on this issue with asterisk config but in realtime it doesn’t supported static peers with version 1.6.2.14. *From:* Ricardo Carvalho [mailto:rjcarvalho.li...@gmail.com] *Sent:* Wednesday, February 16, 2011 10:21 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* Faisal Hanif *Subject:* Re: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP Isn't this a limitation that can be surpassed with some configuration that I'm lacking in my sip.conf or extensions.conf of my asterisk? Ricardo. On Wed, Feb 16, 2011 at 4:54 PM, Faisal Hanif fai...@vopium.com wrote: Well a quick n easy fix for you is you can configure you call sending peers to use username secret in INVITE. As far as I know it possible in almost all CISCO, Avaya and all other standard Gateway and SBCs which follows full SIP RFCs. If you can’t do it then you need to use curl as realtime engine instead of MySQL. It will call a URL for each SIP request which you can handle with flexibility in your CGI scripts with apache. But be careful as per my experience asterisk 1.6 with curl as realtime engine can handle a max of 120 registration in parallel if registration refresh time is 120 seconds. Faisal Hanif *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ricardo Carvalho *Sent:* Wednesday, February 16, 2011 9:41 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with 407 Proxy Authentication Required, I assume because they don't carry the registered contact registration!!! My SIP contacts have type=friend and all inbound calls not coming from my registered phones fall in the default context without authentication, so that someone in the Internet be able to call freely through the Internet anyone in my server's dial plan. Some ideas? Regards, Ricardo Carvalho. -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Multiple SIP endpoint registrations
You can register multiple end users with only one sip account but asterisk does not support ringing all the registered phones on single account. Whenever a new registration comes, asterisk updates its contact info in memory. So if the registration is coming from multiple end users (multiple ip address and port) then the call will be placed to the phone who sent latest registration request. Asterisk does not keep track of all the ip addresses for single account registration. What we have done to ring all the end users with same account is that we listen to registration requests thru manager api in order to detect multiple registration. If we have detected multiple registration then we store the contact information of all the end user phones which are related to single account. And when asterisk receives a dial request for that user, we create a temporary/fake users (as many as needed) in memory and dial all of them in the code not thru Dial application as it does not support thsi scenario. We are still working on this scenario. It is in working condition but in testing phase. On Wed, Mar 9, 2011 at 4:14 PM, --[ UxBoD ]-- ux...@splatnix.net wrote: Hi, With Asterisk 1.8 is it now possible to register the same SIP account at multiple endpoints and for both to ring when the associated extension is dialed ? -- Thanks, Phil -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Prepaid Billing other than A2Billing
If you're in the market for a custom solution for whatever reason, there's more than a few of us who can write a custom prepaid solution. I've done about 7 so far personally and I know there's more like me out there Yes you are right. I am now one of them (i took the red pill :) On Sun, Mar 6, 2011 at 8:39 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: If you're in the market for a custom solution for whatever reason, there's more than a few of us who can write a custom prepaid solution. I've done about 7 so far personally and I know there's more like me out there On Sat, Mar 5, 2011 at 11:26 AM, bilal ghayyad bilmar...@yahoo.comwrote: Hi All; Any one advise for open source prepaid billing other than A2Billing that can work with Asterisk and it is rich by features (for large business)? Regards Bilal -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] 2 ip phones and 1 normal, can't neither send nor receive calls at all...
If you can post your extensions.conf, sip.conf and features.conf then maybe some one can understand and help with your problem. Thanks On Sat, Mar 5, 2011 at 5:42 AM, Francisco Javier Cintrón Olguín fcintr...@gmail.com wrote: I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco spa8800, all them are internal lines. 1.- spa921, 401 ext 2.- spa921, 402 ext 3.- normal phone connected to spa8800 404 ext. It had a very strange behavior when I was configuring call transfer and call pickup. These are steps to repeat it: 1.- from 401 call to 404 2.- from 404 don't answer it. 3.- from 402 press *8 and wait 10 seconds 4.- 402 says that it is connected. 5.- 404 stops to sound. 6.- 401 keeps ringing 7.- Hang up 402 8.- Hang up 401 After these steps I can not neither send nor receive calls from anyone of 401, 402 or 404 until I restart asterisk. /var/log/asterisk/messages, doesn´t show anything strange. ¿what's happening with my phones? Thank you for your kind help. -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] CDR and call transfers :)
Hi all, I have a problem with CDRs when doing call transfers. I am using * 1.8.2.3 with cdr_odbc. As most of you may already know, CDRs and call transfers dont go along very well in *. I mean the developers team have done their best to bring it to an acceptable level. But still it cannot meet the needs of all of the users, some like me are too Whats happening here is when i transfer a call upto two or three or more times, not all the cdrs are generated all the time, hence if i lose one of them, i lose money. It happens with both attended and blind transfers. Bad news is my system cannot afford to not have call transfers facility. I also created call transfers in dialplan with the combination of call parkling and blind transfer (blind transfer seems to generate correct cdrs most of the times). Anyways it did not work (call transfer worked but CDRs didnt) Now I am working on another plan. I am using the builtin transfer facility of * but I have modified some of the code of these features so that whenever these features execute, they send a manager event stating a transfer occured with the following information: Event: Dial Privilege: call,all Timestamp: 1299577784.825096 SubEvent: *Blind Transfer* *Transferer: SIP/pepsi-0002 Transferee: SIP/coke-0003* UniqueID1: 1299577741.2 UniqueID2: 1299577741.3 LinkedID1: 1299577741.2 LinkedID2: 1299577741.3 Transfer To: 17142545586 Transfer Context: siga-external I plan to watch for this event thru AMI, and record who was invloved in transfers, hopefully correct the bill sec and duration with the help of some other events and their timestamps (UnLink Event, Hangup Event etc) in the cdr after it has been inserted in DB, or if its not there in DB, i will insert my own :) This is the best supposed solution i have come up with. But, I am here to ask you people for your ideas and thoughts on my solution. I am still in search for a better solution. So please share your ideas. Thanks PS. I am sending this message to both users and developers list coz i am not sure where this message truly belongs. -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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-dev] CDR and call transfers :)
Thanks Klaus, Actually I got my idea from CEL. But I am more familiar with AMI, plus CEL generates too many events for a single call. I dont want that, I already have a library of routines which read manager events, i just have to plug in my new idea. But still I will think about CEL once again. I never gave it a second thought. I like your idea of a gateway asterisk as well. Will try it. Thanks On Tue, Mar 8, 2011 at 3:27 PM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Am 08.03.2011 11:05, schrieb Rizwan Hisham: Hi all, I have a problem with CDRs when doing call transfers. I am using * 1.8.2.3 with cdr_odbc. This is the best supposed solution i have come up with. But, I am here to ask you people for your ideas and thoughts on my solution. I am still in search for a better solution. So please share your ideas. Sounds like you are trying to re-implement CEL: https://wiki.asterisk.org/wiki/display/AST/Call+Event+Log+%28CEL%29+Driver+Modules https://wiki.asterisk.org/wiki/display/AST/CEL+Design+Goals klaus PS: I prefer a dedicated GW-Asterisk which does accounting: /---trunk1 / /-trunk2 SIP PBX GW Asterisk ---ISDN phones Asterisk \--ISDN2 \---... So, all the transfers happens in the PBX Asterisk. All calls which will be billed are routed via the GW-Asterisk into the PSTN via several uplinks or back to the same or another PBX Asterisk. So, I generate CDRs only at the GW-Asterisk, and as there never happens any transfers on the GW-Asterisk, those CDRs are always 100% correct (as long as you signal proper CLIs from the PBX Asterisk to the GW-Asterisk). -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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-dev] CDR and call transfers :)
Anymore suggestions please. On Tue, Mar 8, 2011 at 3:36 PM, Rizwan Hisham rizwanhas...@gmail.comwrote: Thanks Klaus, Actually I got my idea from CEL. But I am more familiar with AMI, plus CEL generates too many events for a single call. I dont want that, I already have a library of routines which read manager events, i just have to plug in my new idea. But still I will think about CEL once again. I never gave it a second thought. I like your idea of a gateway asterisk as well. Will try it. Thanks On Tue, Mar 8, 2011 at 3:27 PM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Am 08.03.2011 11:05, schrieb Rizwan Hisham: Hi all, I have a problem with CDRs when doing call transfers. I am using * 1.8.2.3 with cdr_odbc. This is the best supposed solution i have come up with. But, I am here to ask you people for your ideas and thoughts on my solution. I am still in search for a better solution. So please share your ideas. Sounds like you are trying to re-implement CEL: https://wiki.asterisk.org/wiki/display/AST/Call+Event+Log+%28CEL%29+Driver+Modules https://wiki.asterisk.org/wiki/display/AST/CEL+Design+Goals klaus PS: I prefer a dedicated GW-Asterisk which does accounting: /---trunk1 / /-trunk2 SIP PBX GW Asterisk ---ISDN phones Asterisk \--ISDN2 \---... So, all the transfers happens in the PBX Asterisk. All calls which will be billed are routed via the GW-Asterisk into the PSTN via several uplinks or back to the same or another PBX Asterisk. So, I generate CDRs only at the GW-Asterisk, and as there never happens any transfers on the GW-Asterisk, those CDRs are always 100% correct (as long as you signal proper CLIs from the PBX Asterisk to the GW-Asterisk). -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] (fast) AGI and AMI synchronization ?
You can use threads and queues in your program to interface with AMI. Your main thread should get all the events from * and based on some logic enqueue it. Some other thread should be listening to the queue and in that thread you are free to read the input whenever you want. This way you are free to keep listening from the AMI as fast as you can and also your other thread will process them at its own pace. Cheers On Tue, Mar 8, 2011 at 5:27 PM, Faisal Hanif fai...@vopium.com wrote: AMI is single threaded link so waiting on it will bring things to hang mode but FastAGI dialplan is multithread. Better to manage all info by AMI in a local hash or array and use sleep/waiting on AGI till required info populated to hash/array by AMI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Corentin Le Gall Sent: Tuesday, March 08, 2011 4:31 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (fast) AGI and AMI synchronization ? Hi, I've been developing some CTI software around asterisk for a while, mainly with the help of AMI and fast AGI. It works quite fine, but I have some trouble sometimes with the un-synchronized property of these 2. Let me explain, we have a dialplan like this one : exten = s,n,UserEvent(useful_input_data) (...) a few actions exten = s,n,AGI(agi://127.0.0.1:/fetch,queuename) The idea is to setup a cti server that talks with both AMI and AGI channels, the first one mainly when one just want to send some data from asterisk to the cti server, and the second one when the dialplan needs some data from this server. My issue is that the AGI requests are received (from the CTI server point of vue) a little bit before the AMI events. In most cases, I don't really care because it is only a little, and the data asterisk needs to fetch from the AGI are set on time. But sometimes not, especially in cases like above, when there are only a few dialplan lines between UserEvent and AGI ... In order to handle that, I thought let's make a sync/meeting point, with the help of the AMI NewExten event, when the app is AGI. The idea would be to keep the AGI connection open as long as the good AMI NewExten event is not received, then to reply and close it, in order for the dialplan to proceed. However, when trying to do this, nothing more occurs on the AMI connection, thus I come to a deadlock ... My question is then, before switching to -dev issues : is there an option somewhere to handle this, whether on the AMI or on the AGI side ? The asterisk version we've been using for a long time is 1.4 and my current attempts are done on 1.8 branch. Thanks, -- Corentin LE GALL Proformatique (Groupe Avencall) - 10bis rue Lucien Voilin - F-92800 Puteaux Tel (+33/0)1.41.38.99.60 - Fax (+33/0)1.41.38.99.70 http://wiki.xivo.fr -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Failover Routing
You can use exten pattern matching for un allocated numbers, say exten= _X.,1,Goto(somewhere) will match all the numbers on priority 1. But make sure you match full extension numbers first which are allocated. Also this extension is a security risk as well. It is recommended that you use a filter dialplan application/function before matching this extension to make sure you accept numbers only. Cheers On Tue, Mar 1, 2011 at 5:11 PM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: Hi, If I use dialstatus variable, it doesn’t give exact reasons for failure like for unallocated numbers it sends Congestion. Whereas, for unallocated number I don’t want to go to failover routing. But need to go to failover routing for other congestion reasons. So, is there any way to check SIP responses like 4xx or 5xx ? Thanks, Deepika -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] asterisk security....again
Hi all, The problem I have been experiencing since last month is that some of my customers are getting calls with Asterisk Unknown caller id. Most of them in the middle of the night. And my asterisk server has no record of these calls. The customers were getting irritated as you can imagine. I guessed the only way to receive incoming calls by by-passing the registration server is thru sip-uri calls directly to customers. I have updated the customers atas to not accept any calls from sources other than the registration server. Thats all fine now. But the question is how can anyone know the direct sip uri addresses of our customers. My guess is that someone has been sniffing my server's sip traffic. In that case what should i do to get rid of the sniffers? If you think there is another reason for that then please tell me even if you dont have the solution. Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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 security....again
thanks for the replies. I dont want to rule-out the possibility of network sniffing. I am sure its not an inside job. The server is off-site and is hosted by a very well reputed hosting company. So if someone is sniffing, what should I do? Probably, you are receiving INVITE attacks from some tool like sipvicious. You should rearange your network to cover some inportant security issues. I have tested sipvicious against my asterisk server already, its been secured that way. Probably your network is exposed to the Internet. To address those situations, you can use a distinct VLAN to address SIP phones and you also can use port security at the switching ports where you connect your ATAs and phones. You should also deliver with tagging (802.1Q) that VLAN to those ATAs and phones. This should protect you from inside sniffers. This VLAN should just communicate with the DMZ where you should have your asterisk server and between those two networks you should only open the needed ports - for a common SIP infrastructure you should open UDP 5060 and the specified UDP range shown in rtp.conf file for the media to pass. Phones VLAN should not communicate directlly with the world, just in the outbound direction if you like. I will talk to my network admin about this. I dont have any wireless network interface to our server. And I am going to apply that IP table thing to the server. Any more suggestions please? On Mon, Feb 28, 2011 at 4:31 PM, Ricardo Carvalho rjcarvalho.li...@gmail.com wrote: Probably, you are receiving INVITE attacks from some tool like sipvicious. You should rearange your network to cover some inportant security issues. The IP address of you server can be revealed in some unincrypted SIP signaling of some call through the Internet to/from your server's client, or simply by your client SRV record in the DNS, if you added it to his DNS. Probably your network is exposed to the Internet. To address those situations, you can use a distinct VLAN to address SIP phones and you also can use port security at the switching ports where you connect your ATAs and phones. You should also deliver with tagging (802.1Q) that VLAN to those ATAs and phones. This should protect you from inside sniffers. This VLAN should just communicate with the DMZ where you should have your asterisk server and between those two networks you should only open the needed ports - for a common SIP infrastructure you should open UDP 5060 and the specified UDP range shown in rtp.conf file for the media to pass. Phones VLAN should not communicate directlly with the world, just in the outbound direction if you like. Regards, Ricardo Carvalho. On Mon, Feb 28, 2011 at 10:33 AM, Rizwan Hisham rizwanhas...@gmail.comwrote: Hi all, The problem I have been experiencing since last month is that some of my customers are getting calls with Asterisk Unknown caller id. Most of them in the middle of the night. And my asterisk server has no record of these calls. The customers were getting irritated as you can imagine. I guessed the only way to receive incoming calls by by-passing the registration server is thru sip-uri calls directly to customers. I have updated the customers atas to not accept any calls from sources other than the registration server. Thats all fine now. But the question is how can anyone know the direct sip uri addresses of our customers. My guess is that someone has been sniffing my server's sip traffic. In that case what should i do to get rid of the sniffers? If you think there is another reason for that then please tell me even if you dont have the solution. Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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 security....again
You are right Terry. Sorry i did not describe full scenario before. Yes the users are remote with atas on port 5060. Attacks on the remote customers was my second guess. My network/system admin has already ruled out the implementation of VPN. In summary, we dont want to do anything on remote customer side. All kind of security and attck prevention techniques have to be implemented on the server. Its comforting to hear someone say they are harmless. But still i would like to know their next step of attack after guessing/scanning. Or is it the only step? On Mon, Feb 28, 2011 at 5:32 PM, Terry Brummell te...@brummell.net wrote: When he says “customers” I am assuming he means remote customers. It sounds like he is a reseller of telecom facilities to me. Which means his customers most likely have ATA’s with port 5060 forwarded to the ATA, or they are direct on the I’net. He has already set the ATA to only allow calls from the proxy server, so sounds like he has plugged the hole. They are not ‘sniffing’ your traffic, they are guessing/scanning. That’s it, that’s all, no great conspiracy going on. They look for open 5060, then send SIP requests to it hopefully finding a badly implemented SIP solution to which they can dial through. Once they determine they cannot get through, the script will move on to the next sucker. You have a couple of options, which you could implement at **each** of your customers if you wanted. Set up a VPN, tunnel the SIP/RTP traffic through it. Set up IPTables at the customer to only allow SIP from your IP. Or, do what you have already done and forget about these idiots doing the scan, they are harmless at this point. Vlans and DMZ for the server do no good as the attacks are being directed at the remote client side, not the server. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ricardo Carvalho *Sent:* Monday, February 28, 2011 6:31 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] asterisk securityagain Probably, you are receiving INVITE attacks from some tool like sipvicious. You should rearange your network to cover some inportant security issues. The IP address of you server can be revealed in some unincrypted SIP signaling of some call through the Internet to/from your server's client, or simply by your client SRV record in the DNS, if you added it to his DNS. Probably your network is exposed to the Internet. To address those situations, you can use a distinct VLAN to address SIP phones and you also can use port security at the switching ports where you connect your ATAs and phones. You should also deliver with tagging (802.1Q) that VLAN to those ATAs and phones. This should protect you from inside sniffers. This VLAN should just communicate with the DMZ where you should have your asterisk server and between those two networks you should only open the needed ports - for a common SIP infrastructure you should open UDP 5060 and the specified UDP range shown in rtp.conf file for the media to pass. Phones VLAN should not communicate directlly with the world, just in the outbound direction if you like. Regards, Ricardo Carvalho. On Mon, Feb 28, 2011 at 10:33 AM, Rizwan Hisham rizwanhas...@gmail.com wrote: Hi all, The problem I have been experiencing since last month is that some of my customers are getting calls with Asterisk Unknown caller id. Most of them in the middle of the night. And my asterisk server has no record of these calls. The customers were getting irritated as you can imagine. I guessed the only way to receive incoming calls by by-passing the registration server is thru sip-uri calls directly to customers. I have updated the customers atas to not accept any calls from sources other than the registration server. Thats all fine now. But the question is how can anyone know the direct sip uri addresses of our customers. My guess is that someone has been sniffing my server's sip traffic. In that case what should i do to get rid of the sniffers? If you think there is another reason for that then please tell me even if you dont have the solution. Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com
Re: [asterisk-users] asterisk security....again
Any suggestions on encrypting the sip and rtp. I have done some googling on it. looks like it is not supported by most end point devices or service providers. But still your thoughts will be appreciated on this subject. On Mon, Feb 28, 2011 at 6:13 PM, Rizwan Hisham rizwanhas...@gmail.comwrote: You are right Terry. Sorry i did not describe full scenario before. Yes the users are remote with atas on port 5060. Attacks on the remote customers was my second guess. My network/system admin has already ruled out the implementation of VPN. In summary, we dont want to do anything on remote customer side. All kind of security and attck prevention techniques have to be implemented on the server. Its comforting to hear someone say they are harmless. But still i would like to know their next step of attack after guessing/scanning. Or is it the only step? On Mon, Feb 28, 2011 at 5:32 PM, Terry Brummell te...@brummell.netwrote: When he says “customers” I am assuming he means remote customers. It sounds like he is a reseller of telecom facilities to me. Which means his customers most likely have ATA’s with port 5060 forwarded to the ATA, or they are direct on the I’net. He has already set the ATA to only allow calls from the proxy server, so sounds like he has plugged the hole. They are not ‘sniffing’ your traffic, they are guessing/scanning. That’s it, that’s all, no great conspiracy going on. They look for open 5060, then send SIP requests to it hopefully finding a badly implemented SIP solution to which they can dial through. Once they determine they cannot get through, the script will move on to the next sucker. You have a couple of options, which you could implement at **each** of your customers if you wanted. Set up a VPN, tunnel the SIP/RTP traffic through it. Set up IPTables at the customer to only allow SIP from your IP. Or, do what you have already done and forget about these idiots doing the scan, they are harmless at this point. Vlans and DMZ for the server do no good as the attacks are being directed at the remote client side, not the server. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ricardo Carvalho *Sent:* Monday, February 28, 2011 6:31 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] asterisk securityagain Probably, you are receiving INVITE attacks from some tool like sipvicious. You should rearange your network to cover some inportant security issues. The IP address of you server can be revealed in some unincrypted SIP signaling of some call through the Internet to/from your server's client, or simply by your client SRV record in the DNS, if you added it to his DNS. Probably your network is exposed to the Internet. To address those situations, you can use a distinct VLAN to address SIP phones and you also can use port security at the switching ports where you connect your ATAs and phones. You should also deliver with tagging (802.1Q) that VLAN to those ATAs and phones. This should protect you from inside sniffers. This VLAN should just communicate with the DMZ where you should have your asterisk server and between those two networks you should only open the needed ports - for a common SIP infrastructure you should open UDP 5060 and the specified UDP range shown in rtp.conf file for the media to pass. Phones VLAN should not communicate directlly with the world, just in the outbound direction if you like. Regards, Ricardo Carvalho. On Mon, Feb 28, 2011 at 10:33 AM, Rizwan Hisham rizwanhas...@gmail.com wrote: Hi all, The problem I have been experiencing since last month is that some of my customers are getting calls with Asterisk Unknown caller id. Most of them in the middle of the night. And my asterisk server has no record of these calls. The customers were getting irritated as you can imagine. I guessed the only way to receive incoming calls by by-passing the registration server is thru sip-uri calls directly to customers. I have updated the customers atas to not accept any calls from sources other than the registration server. Thats all fine now. But the question is how can anyone know the direct sip uri addresses of our customers. My guess is that someone has been sniffing my server's sip traffic. In that case what should i do to get rid of the sniffers? If you think there is another reason for that then please tell me even if you dont have the solution. Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
Re: [asterisk-users] asterisk security....again
Thanks Mr. Kevin. Can anyone please also tell me which firewall is best suited for asterisk/sip attack prevention. Is there any firewall built specially to address sip security problems? On Mon, Feb 28, 2011 at 6:38 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/28/2011 07:27 AM, Rizwan Hisham wrote: Any suggestions on encrypting the sip and rtp. I have done some googling on it. looks like it is not supported by most end point devices or service providers. But still your thoughts will be appreciated on this subject. You cannot protect a remote SIP endpoint from attacks via your server; that SIP endpoint is an endpoint itself, and if it can receive IP packets from attackers, it will process them. These packets don't go through your server, and encrypting the legitimate traffic between your server and the remote endpoint isn't going to make any difference at all. The *only* way to address attacks like this is to modify the configuration of the remote endpoint to ignore all incoming packets that aren't from your server(s). Even that is not a perfect solution, though, because the attacker (if they are actually aware of your server and customers) can spoof the IP addresses of your server(s) in order to get the remote endpoints to at least accept an INVITE (they can't place a successful call through them using spoofing though). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Unknown calls
Hi there everyone, I am a bit confused these days due to some problem I am having. Its not a technical problem. Asterisk is working fine. Most of the users are happy, but some handful of users are getting calls in the middle of the night even though they have enabled Anonymous Call Rejection (blocks calls with no caller id on asterisk server) and TIMED DO NOT DISTURB which also blocks calls unconditionally from 11pm to 6 am. Now the seems to have make it through to the user still. The caller id of the call is Asterisk Unknown. about six users are getting this call only at night time. Asterisk server has no record of this call in log file or cdr. I have also blocked all incoming calls coming from unknown ip addresses etc. Still last night there was a call to a customer. Plz help me figure out the solution for this problem. Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] extend the timout on ringing for pri or sip
use the timeout option in the Dial application like so Dial(SIP/trunk,120) If you dont specify the timeout the default timeout used bya sterisk is probably more than 60 seconds. On Wed, Feb 23, 2011 at 3:17 PM, Israel Gottlieb isr...@gmail.com wrote: Hi Does anyone know how i could extend the timer for the ringing time on a pri or sip trunk ? Today the call gets a cancel request after a minute if not answerd yet is it on asterisk or is a provider side setting? -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Unknown calls
Thats what im unsure about. I think the calls maybe going to the user directly through sip uri or some other method. How can i test that. I have already tried to call those customers with direct sip uri dial but does not work. On Thu, Feb 24, 2011 at 3:19 PM, Roger Burton West ro...@firedrake.orgwrote: On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote: Still last night there was a call to a customer. Plz help me figure out the solution for this problem. Can you be sure that the call _is_ coming through your Asterisk server, rather than being the result of random scanning for your customers' phones? -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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 in dialing out
try this http://www.voip-info.org/wiki/view/Asterisk+sip+qualify On Sat, Feb 19, 2011 at 5:00 AM, asterisk asterisk aster...@ck-lee.comwrote: I have a sip trunk connecting to a huawei softx3000. At the moment, I can register and dial in. However, peer status shows not reachable sip show peer as follow * Name : cmphone Secret : Set MD5Secret: Not set Remote Secret: Not set Context : from-cmphone Subscr.Cont. : device-hints Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes Outb. proxy : 202.0.179.3 DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 202.0.179.3 Addr-IP : 202.0.179.3:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 852350xx SIP Options : 100rel Codecs : 0xe (gsm|ulaw|alaw) Codec Order : (alaw:20,ulaw:20,gsm:20) Auto-Framing : No 100 on REG : No Status : UNREACHABLE Useragent: Reg. Contact : Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No In sip.conf I have register = 852350x:secret@202.0.179.3 [cmphone] type = friend host = 202.0.179.3 secret = secret username = 852350x context = from-cmphone dtmfmode = rfc2833 outboundproxy = 202.0.179.3 caninvite=no insecure = port,invite nat = yes When debug is on, the error message is --- SIP read from UDP:202.0.179.3:5060 --- SIP/2.0 504 Server Time-out From: asterisk sip:aster...@sip.x.xxx;tag=as2d14b9ec To: sip:202.0.179.3;tag=6b0704d0 CSeq: 102 OPTIONS Call-ID: 17e0315c21d7dbc10e8c185740e21...@sip.x.xxx Via: SIP/2.0/UDP 14.xxx.xxx.xxx:5060;branch=z9hG4bK3646eaf2;received=14.xxx.xxx.xxx;rport=5060 Content-Length: 0 Any help is appreciate. CK -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Unknown calls
Its a pure VoIP setup. no cards. On Thu, Feb 24, 2011 at 7:12 PM, Satish Patel satish...@hotmail.com wrote: Do you have PRI card or FXO card? -- Sent from my iPhone On Feb 24, 2011, at 5:28 AM, Rizwan Hisham rizwanhas...@gmail.com wrote: Thats what im unsure about. I think the calls maybe going to the user directly through sip uri or some other method. How can i test that. I have already tried to call those customers with direct sip uri dial but does not work. On Thu, Feb 24, 2011 at 3:19 PM, Roger Burton West ro...@firedrake.org ro...@firedrake.org wrote: On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote: Still last night there was a call to a customer. Plz help me figure out the solution for this problem. Can you be sure that the call _is_ coming through your Asterisk server, rather than being the result of random scanning for your customers' phones? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.comrizwanhas...@gmail.com W: http://www.axvoice.com/www.axvoice.com -- _ -- 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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] Carrying context from one server to another?
you can also set some kind of authentication on the extensions for example ask for a pin to dialout. etc On Thu, Feb 24, 2011 at 6:51 PM, Daniel Tryba dan...@tryba.nl wrote: On Thu, Feb 24, 2011 at 11:38:17AM +, Roger Burton West wrote: The relevant part of my setup is something like: SIP phones - local server - remote server - SIP-to-PSTN provider I want _some_ of the SIP phones on the local server to be able to get access to SIP-to-PSTN, but not all of them. The local-to-remote connection is IAX2 over VPN. The way I would to this is by blocking them on the localserver (with different contexts). An other solution would be to set prefixes on the extension when dialing from local to remote and use these to filter, not very elegant but works over any transport. I use this to do multitenant billing on the remote server in places where I only want 1 IAX trunk. Whether this is effective depends on your control of the local server. -- Daniel Tryba -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] phoneprov
Hi List, Can anyone please tell me how to use the phoneprov.conf to provision my client's atas. I read the file but dont know how to actually use it. -- Best Regards Rizwan Qureshi -- _ -- 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] clustering
anymore ideas anyone please? On Fri, Oct 15, 2010 at 8:36 PM, Josef Grand josef.gra...@gmail.com wrote: use camailio for SIP SLB sip load balancer - Original Message - *From:* Rizwan Hisham rizwanhas...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Thursday, October 14, 2010 5:01 PM *Subject:* [asterisk-users] clustering Hi all, I am planning to do clustering for my company's asterisk servers. I dont know much about it, just read some articles on the internet and learned how to use DUNDi and some basic information about clustering. What I need to know is: 1. can i register end user with multiple asterisk servers at a time? 2. If not, Can I re-route registeration requests to different servers using 1 asterisk server as a gateway and multiple clustered asterisk servers behind it? cheers Thanks in advance -- Best Regards Rizwan Qureshi -- -- _ -- 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 __ Information provenant d'ESET NOD32 Antivirus, version de la base des signatures de virus 5531 (20101014) __ Le message a iti virifii par ESET NOD32 Antivirus. http://www.eset.com __ Information provenant d'ESET NOD32 Antivirus, version de la base des signatures de virus 5531 (20101014) __ Le message a été vérifié par ESET NOD32 Antivirus. http://www.eset.com -- _ -- 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 -- Best Regards Rizwan Qureshi -- _ -- 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] clustering
Hello Zeeshan, How about doing the mixture of what I want to do with your strategy. I mean, what if we have 3 asterisk servers with distributed registrations and also have heartbeat installed monitoring all the servers? will that work? On Mon, Oct 18, 2010 at 9:34 PM, Zeeshan Zakaria zisha...@gmail.com wrote: How about setting up a high availability cluster using DRBD and Heartbeat? There is some good info on it on the Internet. In this type of setup you have two exact same servers running in parallel, and only one has the required services up. They keep themselves in sync. When the primary one goes down, the secondary instantly takes over. Active calls are though dropped, but after that everything is back to normal. There are various other options regarding which server will stay primary, or how and which services will be used on which server. Another option I am exploring is using the same thing but in Proxmox with DRBD. Somebody told me it could be setup so that even the active calls are not dropped. I haven't set it up yet, but will try it when get time. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-18 10:59 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham Sent: Monday, October 18, 2010 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] clustering Unfortunately we are too late to switch to Kamailio. I mean we have developed our pbx with call features and routing on asterisk only. If we switch to some other software that means we will have to redo a lot of development again. I was thinking of using DUNDi and distributing the registrations on different servers. I just dont get one point. lets say if i have 2 users registered on different asterisk servers and... snip Sorry for second post, but I have a Polycom 501 registered to 3 servers. I hit the line button and if the server I pick is down, I don’t get a dial tone. Hope this is useful. -- _ -- 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 -- Best Regards Rizwan Qureshi -- _ -- 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] clustering
Hi all, I am planning to do clustering for my company's asterisk servers. I dont know much about it, just read some articles on the internet and learned how to use DUNDi and some basic information about clustering. What I need to know is: 1. can i register end user with multiple asterisk servers at a time? 2. If not, Can I re-route registeration requests to different servers using 1 asterisk server as a gateway and multiple clustered asterisk servers behind it? cheers Thanks in advance -- Best Regards Rizwan Qureshi -- _ -- 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] Difference
Thanks for sharing all of your thoughts and information. If anyone knows a good article about asterisk 1.8 then please let me know about it. I have read the presentation by Kevin Fleming but more information is always good. Cheers On Wed, Oct 6, 2010 at 10:28 AM, Miguel Molina mmol...@millenium.com.cowrote: I find 1.6.2.13 version is stable for trunk call routing, and it should be too for basic call center use. The asterisk team has made some architectural improvements (moving to astobj2 a lot of internal structures, and much more you may not see from a user perspective) but given the several environment and different use cases, fear to upgrade or proven 1.4 stability for the job, the people usually don't upgrade or make it slowly with a lot of previous tests before making the jump. If you use FAX, I recommend you 1.6.2 or later. The app_fax module is far better than the ast-agx-addons for 1.4. The good old (now unsupported) 1.2 works for many people, ask Steve. So it's up to you. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center El 06/10/10 11:04, Zeeshan Zakaria escribió: For a production environment, 1.4 is the most stable, and it has everything one needs to setup a telecom platform. As per my understanding 1.6 never got the same recognition for stability as 1.4, plus it doesn't have any significant advantages over 1.4. The newer version 1.8 series might be my next jump once it'll be out of beta, but at this time it should not be used in a production environment. Many of us still use 1.4 in production and if you are just starting, this'll be your best choice. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-06 11:54 AM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Be... *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham *Sent:* Wednesday, October 06, 2010 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Difference Is there any major architectural difference between 1.4 and 1.8? The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from , to |) and the AGI structure is enhanced. If you don’t use AGI’s, a qualified “not really”. -- _ -- 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 -- Best Regards Rizwan Qureshi -- _ -- 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] Alert-Info advice
I use the following syntax for sipura i think, and it works fine for me. exten= s,1010,SipAddHeader(Alert-Info: http://127.0.0.1/Bellcore-r2) exten= s,1020,SipAddHeader(Alert-Info: http://127.0.0.1/Bellcore-r3) exten= s,1030,SipAddHeader(Alert-Info: http://127.0.0.1/Bellcore-r4) exten= s,1040,SipAddHeader(Alert-Info: http://127.0.0.1/Bellcore-r5) exten= s,1050,SipAddHeader(Alert-Info: http://127.0.0.1/Bellcore-r6) On Wed, Sep 29, 2010 at 12:38 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! Just out of interest, have you ever got this working? Yes, sure. Mine just isn't but I'm starting to think that my mp3 to 8000Hz Mono 16 bit wav files is a bit dodgy Very well possible. Also look at the individual identity x configuration and consider to select Custom ringtone, then enter the URL for the wav file in question. That is a good and easy way to test if that specifc wav file actually works - also check the http log and/or the snom log. Philipp -- _ -- 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 -- Best Regards Rizwan Qureshi -- _ -- 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] MYSQL ADDON INSTALLATION ERROR
Hi All, Please refresh my memory. I am trying to install asterisk after 2 years. I hav'nt used it since 2008 (version 1.4.2). Now I am trying to install 1.8.0-rc2 on centos 5.5 but getting the following errors. app_mysql.c:33:25: error: mysql/mysql.h: No such file or directory app_mysql.c: In function ‘mysql_ds_destroy’: app_mysql.c:135: warning: implicit declaration of function ‘mysql_close’ app_mysql.c:138: warning: implicit declaration of function ‘mysql_free_result’ app_mysql.c: In function ‘aMYSQL_connect’: app_mysql.c:319: error: ‘MYSQL’ undeclared (first use in this function) app_mysql.c:319: error: (Each undeclared identifier is reported only once app_mysql.c:319: error: for each function it appears in.) app_mysql.c:319: error: ‘mysql’ undeclared (first use in this function) I think i have seen these errors before and did manage to get rid of them but I cant remember how i did it and even dont remember the reason for these errors. Looks like a header file for mysql addon is missing which is actually missing (i have checked). How am I suppose to find it? Plz help. -- Best Regards Rizwan Qureshi -- _ -- 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] MYSQL ADDON INSTALLATION ERROR
Thank you all. It is now installed. On Wed, Oct 6, 2010 at 5:04 PM, Steve Howes steve-li...@geekinter.netwrote: On 6 Oct 2010, at 11:35, Rizwan Hisham wrote: Hi All, Please refresh my memory. I am trying to install asterisk after 2 years. I hav'nt used it since 2008 (version 1.4.2). Now I am trying to install 1.8.0-rc2 on centos 5.5 but getting the following errors. snip Plz help. You need mysql-devel. You might also find that most things are case sensitive, maybe your malfunctioning caps-lock is causing problems? ;) S -- _ -- 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 -- Best Regards Rizwan Qureshi -- _ -- 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] Difference
Hi All, Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk versions. Thanks -- Best Regards Rizwan Qureshi -- _ -- 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] Difference
Is there any major architectural difference between 1.4 and 1.8? On Wed, Oct 6, 2010 at 7:17 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham *Sent:* Wednesday, October 06, 2010 7:15 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Difference Hi All, Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk versions. Thanks -- Best Regards Rizwan Qureshi In a nutshell, 1.4 is the oldest and most stable, 1.6 is the current and 1.8 is the beta version of Asterisk. This is a gross over-simplification, but if you “know nothing”, 1.4 is going to give you the fewest headaches and if you “have to have the latest” 1.6 or 1.8 is the way to go. The ChangeLogs on Asterisk.org will give you a detailed difference. -- _ -- 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 -- Best Regards Rizwan Qureshi -- _ -- 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] Difference
Back in the days i heard that they have changed the architecture in 1.6 and its a lot better than 1.4 (6 times better call handling and robust architecture, someone told me). If they have decided to take the 1.6 architecture to the next level in the new 1.8 version then its a good thing. On Wed, Oct 6, 2010 at 9:58 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 6 Oct 2010, Rizwan Hisham wrote: Is there any major architectural difference between 1.4 and 1.8? Nope. The developer's just got tired of typing .4 Of course, the joke's on them -- 1.8 is only .4 better than 1.4. -- 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 -- Best Regards Rizwan Qureshi -- _ -- 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] MacroExclusive crashed asterisk
Hi all, I think running the macroexclusive application if it is run after hangup (on h extension) crashes asterisk. This has happened a lot of times since i started using the macro exclusive application. There is a situation in my dialplan when after the user hangsup the call, i execute the macro exclusive application and in that macro i disconnect the mysql connection which was made during that call. And asterisk crashes on that extension most of the time (not all the time though). Are we supposed to not use the macroexclusive application on dead or hungup channels? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two sip listening ports for single asterisk
Hi guys, I told my network admin to do what was advised in this thread. It works very well for incoming calls but outgoing calls hangup exactly after 20 secs everytime while displaying the following message on cli: v[Nov 19 10:26:26] WARNING[18617]: chan_sip.c:1910 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) [Nov 19 10:26:26] WARNING[18617]: chan_sip.c:1927 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. == Spawn extension (macro-rating, s, 104) exited non-zero on 'SIP/saad-a8b83300' in macro 'rating' == Spawn extension (macro-rating, s, 104) exited non-zero on 'SIP/saad-a8b83300' Also, this happens only with certain network conditions. in my office the outgoing call hangsup everytime but when i dial from home, both incoming and outgoing calls are fine. So i guess the problem is with user network configuration. asterisk for other users who are using different port to register is listening without any problems. here is my network scenario(from where i make call): we have a zyxel dsl modem connected to our ISP line then we have a D-Link switch connected to the dsl modem then we have sipura 2100 connected to that switch IP addresses are in the range of 192.168.0.0 NAT type on router = SUA ONLY On Thu, Nov 20, 2008 at 5:16 PM, Matthew J. Roth [EMAIL PROTECTED] wrote: Mike wrote: I tried using this iptables sample, and did not see duplicate packets on '--to-ports' port Has some verified this is working for them? I listened on both ports with tcpdump command. Mike, I can confirm that it's working. Admittedly, I never looked at the packets with tcpdump because this *just worked* for me. Calls that were sent to both ports (5060 and 5062) made it to Asterisk which was only listening on port 5060. What's your experience with actual calls? As the original poster, I understand if you want third-party verification. I *thought* this was a slamdunk but I'm not an iptables guru so I'd like it, too. What does the output of iptables-save and lsmod look like? Here's mine, trimmed for relevancy: [EMAIL PROTECTED] ~]# iptables-save # Generated by iptables-save v1.3.5 on Thu Nov 20 12:03:21 2008 *nat :PREROUTING ACCEPT [5579:1727747] :POSTROUTING ACCEPT [1943:176116] :OUTPUT ACCEPT [1943:176116] -A PREROUTING -i eth2 -p udp -m udp --dport 5062 -j REDIRECT --to-ports 5060 COMMIT # Completed on Thu Nov 20 12:03:21 2008 [EMAIL PROTECTED] ~]# lsmod Module Size Used by ip_conntrack_netbios_ns36033 0 ipt_REDIRECT 35009 1 xt_tcpudp 36417 1 iptable_nat40773 1 ip_nat 53101 2 ipt_REDIRECT,iptable_nat ip_conntrack 91237 3 ip_conntrack_netbios_ns,iptable_nat,ip_nat nfnetlink 40457 2 ip_nat,ip_conntrack ip_tables 55329 1 iptable_nat x_tables 50377 4 ipt_REDIRECT,xt_tcpudp,iptable_nat,ip_tables Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] two sip listening ports for single asterisk
Hi all, We are planning to shift our sip users from one platform to another. (basically from one asterisk server to another). the problem we are facing is that both asterisk servers are using different ports to listen for sip. and both have live customers on them. provisioning their ata's is not a good option for us coz of our settup. we cant just ask the customers to change their ports for registration (many of them dont know what a port is) Is it possible to make single asterisk server listen on two different ports? if there is any other option we can use, plz inform me about it. Thanx in advance. -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
The fax is originated from a fax machine connected to an ata which supports t38. On Wed, Sep 24, 2008 at 11:54 PM, C F [EMAIL PROTECTED] wrote: On Wed, Sep 24, 2008 at 5:43 AM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, Sorry to interrupt. I need some help regarding fax passthru mode. We are trying to configure fax passthru mode in asterisk using sip. For out of network calls/fax we use trunk configuration. i am using asterisk 1.4.2. The user has to use fax machine connected to their ata and dial the callee number, the call is originated just like a regular voice call. have not defined any special context for sending faxes. Have enabled t38 and canreinvite in peer/user and trunk configuration. But the fax is not going thru. Our service provider does support fax passthru. Following is the trunk and user/peer configuration: They support passthru, and the originating send fax is what? PSTN? or VoIP ATA with t38 support? There has to one that does the t38, if the point where it gets converted to VoIP does not support t38 then passthru will not help you. TRUNK CONF [TRUNK-OUT] type=peer host=XXX port=5060 context=default country=us dtmfmode=rfc2833 restrictcid=no canreinvite=yes insecure=no disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm promiscredir=yes t38_udptl=yes USER/PEER [abc] username=abc type=friend secret=123 qualify=25000 nat=yes mailbox=12129339037 insecure=port,invite incominglimit=2 outgoinglimit=2 intl_trunk=TRUNK-OUT local_trunk=TRUNK-OUT host=dynamic dtmfmode=inband context=uscan canreinvite=yes callerid=Rizwan Qureshi 122 accountcode=1:0:abc amaflags=default disallow=all allow=ulaw allow=alaw allow=gsm t38_udptl=yes Any solutions? On Tue, Sep 23, 2008 at 10:21 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro [EMAIL PROTECTED] wrote: ATAs work OK I guess, just make sure to use a loss less codec such as ULAW. Since the OP stated he is using E1 lines then he should probably be using alaw instead. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk mysql CDR
You can use the ResetCDR() application with the w flag in it after you get
the unavailable, busy or etc message from the callee. It will store the cdr
of that call and after forwarding to mobile, that cdr will be dumped again.
On Wed, Sep 24, 2008 at 8:26 AM, Nhadie [EMAIL PROTECTED] wrote:
hi,
i'm using this macro to dial an extension and forward to a mobile if
unavailable,busy or noanswer
exten = 100,1,Macro(dial-ext|SIP/${EXTEN}|vm-100|moh-100)
exten = 100,2,Goto(100-${DIALSTATUS}|1)
exten = 100-BUSY,1,Macro(dialout-local-mobile|91234567)
exten = 100-BUSY,2,Voicemail([EMAIL PROTECTED]|u)
exten = 100-CONGESTION,1,Macro(dialout-local-mobile|91234567)
exten = 100-CONGESTION,2,Voicemail([EMAIL PROTECTED]|u)
exten = 100-NOANSWER,1,Macro(dialout-local-mobile|91234567)
exten = 100-NOANSWER,2,Voicemail([EMAIL PROTECTED]|u)
my prob is on the CDR, from extension 500 i called 100, 100 is not
online so it should forward it to my mobile
but on the cdr it shows like this:
FromTo
500 100-CHANUNAVAIL
should it be like
FromTo
500 91234567
or
FromTo
100 91234567
any idea how to fix those?
regards,
nhadie
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--
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Rizwan Hisham
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Re: [asterisk-users] Fax with asterisk
Hi all, Sorry to interrupt. I need some help regarding fax passthru mode. We are trying to configure fax passthru mode in asterisk using sip. For out of network calls/fax we use trunk configuration. i am using asterisk 1.4.2.The user has to use fax machine connected to their ata and dial the callee number, the call is originated just like a regular voice call. have not defined any special context for sending faxes. Have enabled t38 and canreinvite in peer/user and trunk configuration. But the fax is not going thru. Our service provider does support fax passthru. Following is the trunk and user/peer configuration: TRUNK CONF [TRUNK-OUT] type=peer host=XXX port=5060 context=default country=us dtmfmode=rfc2833 restrictcid=no canreinvite=yes insecure=no disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm promiscredir=yes t38_udptl=yes USER/PEER [abc] username=abc type=friend secret=123 qualify=25000 nat=yes mailbox=12129339037 insecure=port,invite incominglimit=2 outgoinglimit=2 intl_trunk=TRUNK-OUT local_trunk=TRUNK-OUT host=dynamic dtmfmode=inband context=uscan canreinvite=yes callerid=Rizwan Qureshi 122 accountcode=1:0:abc amaflags=default disallow=all allow=ulaw allow=alaw allow=gsm t38_udptl=yes Any solutions? On Tue, Sep 23, 2008 at 10:21 PM, Andrew Joakimsen [EMAIL PROTECTED]wrote: On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro [EMAIL PROTECTED] wrote: ATAs work OK I guess, just make sure to use a loss less codec such as ULAW. Since the OP stated he is using E1 lines then he should probably be using alaw instead. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extension definition
You maybe using wrong username. If the user is defined in sip, you should be able to register using the correct username and password. Also, see if asterisk is listening on a defferent sip port instead of default 5060. If its different use that port. On Wed, Sep 24, 2008 at 3:32 AM, michel freiha [EMAIL PROTECTED] wrote: Hello Eric, i didwhat you asked me to do but i'm getting Notfound sip message when trying to register regrads On Tue, Sep 23, 2008 at 9:56 PM, Eric ManxPower Wieling [EMAIL PROTECTED]wrote: This is done in sip.conf, iax.conf, etc, not in extensions.conf. By the time a call gets to extensions.conf it must already be authenticated. Assume the username is robertdobbs and the ip is 209.17.71.61 In sip.conf you would have something like this: [robertdobbs] deny=0.0.0.0/0 permit=209.17.71.61 http://0.0.0.0/0permit=209.17.71.61 rest of the options here michel freiha wrote: Hi all, I need please the exact extension definition under extensions.conf that accepts any call coming from an appropriate username and Ip address...This mean that the authentication should be done on username and IP address Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and prepaid billing
We have done it too. www.axvoice.com On Tue, Sep 23, 2008 at 3:39 PM, Benjamin Jacob [EMAIL PROTECTED]wrote: Hi Bilal, Yes it is definitely possible. And I've done it myself for a couple of our clients. Does that answer your two questions? cheers - Ben. --- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote: From: bilal ghayyad [EMAIL PROTECTED] Subject: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com Date: Tuesday, September 23, 2008, 9:52 AM Hi All; Did anyone do an prepaid billing application via AGI? I would like to know if that is possible. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wad happen if there is nothing wrong in conf but still can't make calls?
you must share your configuration with us. otherwise we cant even make a wild guess. On Mon, Sep 22, 2008 at 7:48 PM, Cindy Tan [EMAIL PROTECTED] wrote: may i noe wad can i do because my asterisk is working fine but the calls cannot proceed between 2 asterisk servers. hope anyone can help me solve this major problem. thanks a lot in advance Regards -- Make the most of what you can do on your PC and the Web, just the way you want. Windows Live http://www.get.live.com/wl/all ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rxfax and txfax
Hi all, I want to configure my asterisk for sending and receiving faxes. I see in my sip.conf that i have to enable the t.38 capability. I have done that but the rxfax and txfax applications are not installed. They are not listed in applications when i do make menuselect. i have searched in voip-info wiki, found a pagehttp://www.voip-info.org/wiki/index.php?page_id=2583tk=99b8d086f0f28f4c1542comments_page=1but the links given on that page for downloading the applications are not working. I am using asterisk 1.4.2, i thaught the missing applications maybe included in latest version of asterisk but they are not, already downloaded and checked in asterisk 1.4.21. How can i install these applications. Are there anyother components required to make my asterisk a fax-passthru system. -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP URI Forwarding
Hi all, I am having a problem with sip uri incoming calls. I have 2 asterisk servers both are 1.4.2. i dial sip uri from one asterisk server which sends the call to the other asterisk server by seeing its domain name in the uri. Invite reaches the recieving asterist server but the call is not autenticated. Everytime i see the following NOTICE on the asterisk server (caller end) [Sep 17 15:38:24] NOTICE[4594]: chan_sip.c:11968 handle_response_invite: Failed to authenticate on INVITE to 'rizwan sip:[EMAIL PROTECTED]:9860 ;tag=as089d4adb' My dialplan on caller end is: [directcall] exten= 123,1,Dial(SIP/abc:[EMAIL PROTECTED]:9060) exten= 123,2,Hangup() exten= 456,1,Dial(SIP/adf:[EMAIL PROTECTED]:9060) exten= 456,2,Hangup() SIP general settings on receiving end are: [general] context=uricall-incoming allowoverlap=no bindport=9060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm relaxdtmf=yes useragent=Asterisk PBX dtmfmode = rfc2833 nat=no canreinvite=yes peer settings on receiving end: [adf] username=adf type=friend secret=XXX qualify=25000 nat=yes insecure=port,invite host=dynamic dtmfmode=rfc2833 context=sipuri-incoming canreinvite=yes callerid=adf xyz 123 accountcode=6:0:adf amaflags=default disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm am i doing something wrong here? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dtmf passthru
hi all, Is there an option of dtmf passthru mode in asterisk. If yes, how can i do it? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf passthru
Alex, So u mean DTMF is by default passed thru already. Benjamin, Well i dont want to enforce my asterisk dtmf setting on any call. What i want is whatever the user has set the dtmf mode in his ata or softphone, that should be used to pass the dtmf signals on to the callee. On Wed, Sep 17, 2008 at 6:05 PM, Alex Balashov [EMAIL PROTECTED]wrote: What is DTMF passthru? DTMF is regenerated by default. If the DTMF mode is inband, it's simply part of the audio stream. If it uses named RTP events, those are regenerated on the other call leg. Rizwan Hisham wrote: hi all, Is there an option of dtmf passthru mode in asterisk. If yes, how can i do it? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP URI Forwarding
thats what i am passing exten= 456,1,Dial(SIP/adf:[EMAIL PROTECTED]:9060 adf is username and 123 is the password On Wed, Sep 17, 2008 at 6:01 PM, Alex Balashov [EMAIL PROTECTED]wrote: If there is a secret= on the receiving peer, the sending peer needs to provide that secret. Along with a username. Rizwan Hisham wrote: Hi all, I am having a problem with sip uri incoming calls. I have 2 asterisk servers both are 1.4.2. http://1.4.2. i dial sip uri from one asterisk server which sends the call to the other asterisk server by seeing its domain name in the uri. Invite reaches the recieving asterist server but the call is not autenticated. Everytime i see the following NOTICE on the asterisk server (caller end) [Sep 17 15:38:24] NOTICE[4594]: chan_sip.c:11968 handle_response_invite: Failed to authenticate on INVITE to 'rizwan sip:[EMAIL PROTECTED]:9860 http://sip:[EMAIL PROTECTED]:9860;tag=as089d4adb' My dialplan on caller end is: [directcall] exten= 123,1,Dial(SIP/abc:[EMAIL PROTECTED]:9060 http://abc:[EMAIL PROTECTED]:9060) exten= 123,2,Hangup() exten= 456,1,Dial(SIP/adf:[EMAIL PROTECTED]:9060 http://adf:[EMAIL PROTECTED]:9060) exten= 456,2,Hangup() SIP general settings on receiving end are: [general] context=uricall-incoming allowoverlap=no bindport=9060 bindaddr=0.0.0.0 http://0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm relaxdtmf=yes useragent=Asterisk PBX dtmfmode = rfc2833 nat=no canreinvite=yes peer settings on receiving end: [adf] username=adf type=friend secret=XXX qualify=25000 nat=yes insecure=port,invite host=dynamic dtmfmode=rfc2833 context=sipuri-incoming canreinvite=yes callerid=adf xyz 123 accountcode=6:0:adf amaflags=default disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm am i doing something wrong here? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] strange transfer problem
Hi all, I am having a strange problem with my asterisk server. When i dial an outside tollfree number, if there is a menu for example press 1 for support, press 2 for sales etc, after pressing any given option as the system begins to transfer me the call hangs up. I have tried it so many times on different tollfree numbers but the problem remains only when i dial from my asterisk box. The call is disconnected with a normal call clearing (hangup cause 16). But when i dial from my cell phone or any other line, the call is transfered without any problem. I also checked sip debug and core debug for different calls. I think it has something to do with strict routing, the only strange message i get on the cli is: [Sep 4 09:10:39] DEBUG[14929]: chan_sip.c:5690 reqprep: Strict routing enforced for session [EMAIL PROTECTED] This message does not appear for other calls (when there is no transfering) I googled a little on strict and loose routing but i did not get it. maybe someone here can help me solve this problem. VERSIONS asterisk 1.4.2 zaptel and libpri 1.4.0 I can send you core debug if you want it. -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callfiles/manager api originate call fails
Actually both calls have to be originated to the outside world. Thats why im using @TRUNK-OUT, when the first call is answered only then the call goes to a context. Thats where the problem is, the first call does not originate so i cant throw it to any context. On Thu, Aug 21, 2008 at 8:47 PM, Anthony Francis [EMAIL PROTECTED]wrote: Rizwan Hisham wrote: Hi all, asterisk is giving me tough time. its been 3 days I am trying to originate outgoing call using manager api/callfiles. I would say remove the @TRUNK-OUT part and make sure that the context you send the call to knows about sending calls to the outside world. -- Thank you and have any kind of day you want, Anthony Francis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] callfiles/manager api originate call fails
Hi all, asterisk is giving me tough time. its been 3 days I am trying to originate outgoing call using manager api/callfiles. both seem to work fine when i originate a call for a local peer, but if i try originating a call outside using a trunk thats when everything goes wrong. It does originate the call but the call does not go through to the desired endpoint. The trunk configuration is correct as all the other calls from users are fine. Am here for any suggestion. How can i make it work. If anyone knows anyother technique to originate auto calls from asterisk i'll be happy to try them out. I am using the following manager command, fputs($socket, Action: Originate\r\n); //fputs($socket, Channel: SIP/abc\r\n); fputs($socket, Channel: SIP/.$txt_your_number.@TRUNK-OUT\r\n); fputs($socket, Context: webcall\r\n); fputs($socket, Exten: 932\r\n); fputs($socket, Priority: 1\r\n); fputs($socket, CallerID: WebCall932\r\n); fputs($socket, Timeout: 3\r\n); fputs($socket, Variable: ID= . $id . |accountcode=7:0:webcall|sec= . $min . |dialnum= . $txt_to_number . |source_num= . $txt_your_number . |calldate= . date(Y-m-d H:i:s) . \r\n\r\n); and my callfile contents are: Channel: SIP/TRUNK-OUT/$DIALNUM CallerID: Webcall932 MaxRetries: 2 RetryTime: 10 WaitTime: 35 Account: 7:0:webcall Context: webcall Extension: 932 Priority: 1 Set: ID=.$id. Set: accountcode=7:0:webcall Set: sec=.$allowed_secs. Set: dialnum=.$dialnum.\ et: source_num=.$srcnum. Set: calldate=.$calldate. .$calltime.\n; -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] shared mysql connection in dialplan
have done it, and its working fine. but still expecting to receive some new ideas. On Wed, Aug 6, 2008 at 2:12 PM, Rizwan Hisham [EMAIL PROTECTED]wrote: hi all, i just finished developing some incoming call features in a macro. that macro gets executed everytime an incoming call is received and a new mysql connection is made using the MYSQL cmd in dialplan. i want to use a single mysql connection for every incoming call. my idea of doing it is like this, i want to get a mysql connection in a global variable, just to share the connection with different incoming calls. Im not sure if this can be done. I am going to try doing it somehow, meanwhile i want your suggestions about how i can share a mysql connection with different calls in a dialplan. I am using asterisk1.4.2 and asterisk addon1.4.0 package for mysql connectivity. Thanx in advance -- Best Regards Rizwan Hisham -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] shared mysql connection in dialplan
hi all, i just finished developing some incoming call features in a macro. that macro gets executed everytime an incoming call is received and a new mysql connection is made using the MYSQL cmd in dialplan. i want to use a single mysql connection for every incoming call. my idea of doing it is like this, i want to get a mysql connection in a global variable, just to share the connection with different incoming calls. Im not sure if this can be done. I am going to try doing it somehow, meanwhile i want your suggestions about how i can share a mysql connection with different calls in a dialplan. I am using asterisk1.4.2 and asterisk addon1.4.0 package for mysql connectivity. Thanx in advance -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help with implementing prepaid in asterisk
You can calculate users remaining minutes according to his remaining balance and then set the Absolute timeout for his every outgoing call using the Timeout(absolute) = X http://www.voip-info.org/wiki/index.php?page=Asterisk+func+timeoutvariable. This will hangup the call after X seconds And if there users balance is in negative or zero then just bypass the dial statement in your dialplan and land on Hangup or Playback before hangup to let the user know why his call is not being connected. All of these changes you have to make in extensions.conf On Tue, Jul 29, 2008 at 11:59 AM, Ian Coetzee [EMAIL PROTECTED]wrote: Hi all I am trying to implement a prepaid dialing system on our asterisk box. I however have a few questions I need to ask. I have written a simple script in php to do all the billing. 1. What do I need to user to cut off the users in mid call 2. What do I need to insert into my dialplan to deny a user to call if you need any config files I will send them, seeing as I dont know what files to send. If you can point me to a howto I will be more gratefull. Regards Ian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Overlap dialing via SIP
i can see from your dialplan that all the extensions except international
extension are of 12 digits. International extensions are of 13 or more
digits. here is what you can do with the international extensions, all other
extensions remain the same:
[084x]
exten = _9084,1,Macro(dialout-pstn)
[outbound-national]
exten = _90[1-2]X,1,Macro(dialout-pstn)
[087x]
exten = _9087,1,Macro(dialout-pstn)
[0906]
exten = _90906XXX,1,Macro(dialout-pstn)
[outbound-international]
exten = _900XX.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _900XX.,2,Congestion
If you see closely i have put a dot at the end of each international
extension, this will allow you to dial atleast 13 digits. so no need to
crate extension of every length.
On Mon, Jul 21, 2008 at 9:10 PM, Ben Thompson [EMAIL PROTECTED] wrote:
Hi
I have set up an asterisk system which allows the use of Overlap Dialing
from
SIP handsets. In order to do this I had to list the various patterns of
numbers
which can be dialed in the UK. We also dial with a prefix of '9' for and
outside
line so much of my dialplan looks like this :-
[084x]
exten = _9084,1,Macro(dialout-pstn)
[outbound-national]
exten = _90[1-2]X,1,Macro(dialout-pstn)
[087x]
exten = _9087,1,Macro(dialout-pstn)
[0906]
exten = _90906XXX,1,Macro(dialout-pstn)
...
I was able to download the mappings for 0800 numbers and other special
ranges
from the ofcom website and I have incorporated these. For international
dialing
I have not been able to find an easy way of doing this so I created the
folling
contexts whcih make use of the WaitExten feature :-
[outbound-international]
exten = _900XX,1,Set(oldexten=${EXTEN})
exten = _900XX,2,Goto(international-number-length-check,s,1)
[international-number-length-check]
exten = s,1,Answer
exten = s,2,WaitExten(8)
exten = _X,1,Set(enddigits=${EXTEN})
exten = _X,2,NoOp(${TIMESTAMP} ok 13 digits - we dial
${oldexten}${enddigits})
exten = _X,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _X,4,Congestion()
exten = _X,104,Busy()
exten = _XX,1,Set(enddigits=${EXTEN})
exten = _XX,2,NoOp(${TIMESTAMP} ok 14 digits - we dial
${oldexten}${enddigits})
exten = _XX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _XX,4,Congestion()
exten = _XX,104,Busy()
exten = _XXX,1,Set(enddigits=${EXTEN})
exten = _XXX,2,NoOp(${TIMESTAMP} ok 15 digits - we dial
${oldexten}${enddigits})
exten = _XXX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _XXX,4,Congestion()
exten = _XXX,104,Busy()
exten = t,1,Dial(${OUTBOUNDTRUNK}/${oldexten})
exten = t,2,Congestion()
exten = t,102,Busy()
This works fairly well but I have noticed that occasionally the WaitExten
feature does
not seem to catch the first digits if they are dialed too quickly. It is
almost as if
there is a some sort of delay and the thirteenth digit is sometimes missed.
Can anyone suggest why WaitExten might be ocasionally missing a digit or
can anyone think
of a better way of doing this?
Thanks
Ben Thompson
___
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Re: [asterisk-users] Help With dial plan
maybe the user is dialing something other than 3000 and that extension is
not registered on your asterisk. just a wild guess.
On Tue, Jul 22, 2008 at 10:41 AM, James Mutuku [EMAIL PROTECTED] wrote:
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo calls
to trixbox. I am using sip to send the calls from the gateway to trixbox. I
have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten = 3000,1,dial(sip/3000)
exten= 3000,2,answer()
exten = 3000,3,congestion()
exten= 3000,4,hangup()
this works fine. But I when I put it in the form
exten = _3XXX,1,dial(sip/${EXTEN})
exten= _3XXX,2,answer()
exten =_3XXX,3,congestion()
exten= _3XXX,4,hangup()
the call goes into congestion and I get a busy tone. What could I be doing
wrong?
James
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[asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)
Hi all, I have setup an asterisk system which: 1. recieves incoming sip calls 2. ask the caller the number they want to dial, and then dial that number 3. after the caller is done talking and callee hangsup or even if the callee does not answer the phone, the caller is asked for another number to dial. 4. And so onuntill the caller hangsup Everthing above is working fine. But i dont know how to manipulate the cdr so that every outgoing call for he caller should be logged. I have looked into ForkCDR but it seems like it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial command and its g option
just add as many extensions as you want under the Dial command extension keeping the extension number same: exten = s,n,Dial(SIP/100,100,Ttg) exten = s,n,Application here On Thu, Jun 12, 2008 at 3:25 PM, voip crazy [EMAIL PROTECTED] wrote: I need to execute an action after a call is hangup. I just see the command Dial has an option for that, the g option. I configure the dial command as exten = s,n,Dial(SIP/100,100,Ttg) How should I add the line which the command will be executed after the dial command in this example? I don`t how its works, someone could put a example about the way to use it. Thanks you in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Brent, hope your problems go away soon. I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We are using asterisk 1.4.2 for a SIP only based configuration. Currently we have about 200 SIP users which can cause approximately upto 3 simultaneous calls. We are mainly concerned about the performance and stability of asterisk when the load increases. Our server can handle about 100 simultaneous calls having 3Ghz Dual Intl-Xeon Processor with 2GB of ram using G711 codec, and around 30 simultanoeus calls using G729 codec. This we are expecting from the hardware. We are planning to accomodate about 5,000 users on this server. Before the release of 1.6 i heard that its architecture is going to be different from 1.2 and 1.4. Recently i read an article about freeswitchhttp://freeswitch.org/node/117, which explains how its functionality is like asterisk but it can perform better than asterisk due to its architectural differences. The main developer for freeswitch is anthony who also codes for asterisk. He explaines why the architecture of asterisk needs to be changed which requires massive recoding, but nobody took the step to do it. Thats why he started freeswitch on his own to redifine the architecture, so that the performance and reliability of the switch should be better than asterisk. In his article he has already said that freeswitch beats asterisk by a factor of 10. If asterisk architecture is being rewritten in 1.6 to achive the same goal, then we will be happy to use 1.6 instead of shifting the whole system to freeswitch. We dont have any problem or issues with 1.4.2 yet. We are mainly concerned about the its performance when the load increases. If 1.6 is more reliable under heavy loads then we would like to use it. If anyone can put some light on this topic, all i can say is thanx for sharing your thaughts and experiences. On Thu, Jun 5, 2008 at 1:13 AM, Brent Davidson [EMAIL PROTECTED] wrote: Just an update. I tried updating to the newest Rhino Release firmware 1.15 and newest stable driver version 2.2.6. It works OK with zaptel-1.4.9.2 and compiles OK with 1.4.10.1 but when compiled against zaptel 1.4.10.1 Asterisk does not see any zap channels. I'm currently running one branch office with the upgraded firmware, driver, zaptel-1.4.9.2 and Asterisk-1.4.20.1. I'll see how everything goes there and may upgrade the other offices if it works OK. Thanks, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] where did the switch statement come from?
Hi all, I have looked up the applications and function in asterisk but i could not find the help for the switch statement which is used in several places in sample extensions.conf file. i am using asterisk 1.4.2. On voip-info.org the switch statement seems to be used to connect 2 asterisk servers, but i could not find a satisfactory explanation for the this statement. Can anybody help me understand the switch statement? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where did the switch statement come from?
where can i find details about both switch statement and dundi. On Mon, May 19, 2008 at 4:37 PM, Alexander Lopez [EMAIL PROTECTED] wrote: The switch statement allows you to 'include' a context from another machine into your machine. Problems with it was if the other machine was unavailable, or even slow to respond, your machine would hang until it timed out. DUNDI has since replaced the functionality of the switch statement and given you so much more in return -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Rizwan Hisham *Sent:* Monday, May 19, 2008 6:59 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] where did the switch statement come from? Hi all, I have looked up the applications and function in asterisk but i could not find the help for the switch statement which is used in several places in sample extensions.conf file. i am using asterisk 1.4.2. On voip-info.orgthe switch statement seems to be used to connect 2 asterisk servers, but i could not find a satisfactory explanation for the this statement. Can anybody help me understand the switch statement? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems passing variables from a macro
I haven't used ael but in extension.conf whenever we set a channel variable
we use a SET command just like you used it to set the variable MACRO_RESULT.
try using the set command, if still it does not work then try to initialize
the wrongpin=0 before the dial command, or outside the macro.
On Fri, May 16, 2008 at 12:36 PM, Tobias Ahlander [EMAIL PROTECTED]
wrote:
Good day,
I'm using a dial string as follows:
Dial(SIP/${phonenumber},30,grM(screen^${pin})L(${30}[:6]));
When I set a variable in the macro screen, it doesn't get passed back to
the extension from where the dial was called. I can always put the result in
the MySQL database, but that feels a bit overkill... the macro looks as
follows:
macro screen (arg1) {
Wait(0.2);
Read(acceptcall|sounds/pin|7);
if(${acceptcall} = ${arg1}) {
NoOp(connect them);
wrongpin=0;
} else {
Set(MACRO_RESULT=CONTINUE);
wrongpin=1;
}
NoOp(MACRO_RESULT = ${MACRO_RESULT});
}
This is the output from the CLI, and I can see that the wrongpin is set to
1, but when I do a NoOp right after leaving the macro, it says its empty...
-- Executing [EMAIL PROTECTED]:36] Dial(SIP/1003-b7619b78,
SIP/1203|30|grM(screen^1234)L(30[:6])) in new stack
-- Limit Data for this call:
timelimit = 30
play_warning = 6
play_to_caller = yes
play_to_callee = yes
warning_freq = 0
start_sound= (null)
warning_sound = beep
end_sound = beep
-- Called 1203
-- SIP/1203-08d62408 is ringing
-- SIP/1203-08d62408 answered SIP/1003-b7619b78
-- Executing [EMAIL PROTECTED]:1] Set(SIP/1203-08d62408, arg1=1234)
in new stack
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/1203-08d62408, 0.2) in
new stack
-- Executing [EMAIL PROTECTED]:3] Read(SIP/1203-08d62408,
acceptcall|sounds/pin|7) in new stack
-- Accepting a maximum of 7 digits.
-- SIP/1203-08d62408 Playing 'sounds/pin' (language 'en')
-- User entered '1'
-- Executing [EMAIL PROTECTED]:4] GotoIf(SIP/1203-08d62408, 0?5:8)
in new stack
-- Goto (macro-screen,s,8)
-- Executing [EMAIL PROTECTED]:8] Set(SIP/1203-08d62408,
MACRO_RESULT=CONTINUE) in new stack
-- Executing [EMAIL PROTECTED]:9] Set(SIP/1203-08d62408, wrongpin=1)
in new stack
-- Executing [EMAIL PROTECTED]:10] NoOp(SIP/1203-08d62408, Finish
if-screen-32753) in new stack
-- Executing [EMAIL PROTECTED]:11] NoOp(SIP/1203-08d62408,
MACRO_RESULT = CONTINUE) in new stack
-- Executing [EMAIL PROTECTED]:37] NoOp(SIP/1003-b7619b78,
DIALSTATUS:ANSWER) in new stack
-- Executing [EMAIL PROTECTED]:38] NoOp(SIP/1003-b7619b78, wrongpin=)
in
new stack
Is there a good way to pass this variable back to the context connect?
Thanks,
Best regards,
Tobias
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Re: [asterisk-users] Manager API - Setvar not working
same is the case in 1.6, i cant set the variable still. On Thu, May 8, 2008 at 8:43 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote: Hi all, I am using a simple perl script to connect with ast manager api. the script tries to set a channel variable. It extracts the channel name from the events it recieves after dial command. When i try to set the channel variable, asterisk responses with an error saying that the channel does not exist. Can anybody tell me why its doing so, coz i can see on cli that the channel exists. If i try to set the variable without stating the channel then it sets the global variable, but i want to set the channel variable. Anybody has a solution to this problem? In 1.6 you can set a channel variable from a diferent context. I don't think you can do that in 1.4 . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API - Setvar not working
Thanx a lot.that was it. will never forget to remove the new character again. Now its working fine. On Fri, May 9, 2008 at 4:31 PM, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Rizwan Hisham [EMAIL PROTECTED] wrote: same is the case in 1.6, i cant set the variable still. My guess would be that you have a problem with line endings. All lines received from the manager interface are terminated with \r\n, not just \n. If you only strip the \n off, the channel name you received will contain \r at the end. If you then send that back in the setvar command with a further \r\n to terminate the line, then Asterisk will probably not find a matching channel. Make sure you strip both \r and \n from incoming lines. Cheers Tony On Thu, May 8, 2008 at 8:43 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote: Hi all, I am using a simple perl script to connect with ast manager api. the script tries to set a channel variable. It extracts the channel name from the events it recieves after dial command. When i try to set the channel variable, asterisk responses with an error saying that the channel does not exist. Can anybody tell me why its doing so, coz i can see on cli that the channel exists. If i try to set the variable without stating the channel then it sets the global variable, but i want to set the channel variable. Anybody has a solution to this problem? In 1.6 you can set a channel variable from a diferent context. I don't think you can do that in 1.4 . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham -=-=-=-=-=- [Alternative: text/html] -=-=-=-=-=- -=-=-=-=-=- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=- -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API - Setvar not working
Can anybody help in parsing the manager events efficiently? Any ideas? On Fri, May 9, 2008 at 5:07 PM, Gunārs Grundāns [EMAIL PROTECTED] wrote: On 5/8/08, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote: Hi all, I am using a simple perl script to connect with ast manager api. the script tries to set a channel variable. It extracts the channel name from the events it recieves after dial command. When i try to set the channel variable, asterisk responses with an error saying that the channel does not exist. Can anybody tell me why its doing so, coz i can see on cli that the channel exists. If i try to set the variable without stating the channel then it sets the global variable, but i want to set the channel variable. Anybody has a solution to this problem? In 1.6 you can set a channel variable from a diferent context. I don't think you can do that in 1.4 . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My guess that Setvar was deprecated in Asterisk 1.4. Use Set instead if you are trying to use it in 1.6 -- Gunars Grundans http://freight.lv ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager API - Setvar not working
Hi all, I am using a simple perl script to connect with ast manager api. the script tries to set a channel variable. It extracts the channel name from the events it recieves after dial command. When i try to set the channel variable, asterisk responses with an error saying that the channel does not exist. Can anybody tell me why its doing so, coz i can see on cli that the channel exists. If i try to set the variable without stating the channel then it sets the global variable, but i want to set the channel variable. Anybody has a solution to this problem? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI asterisk high balance
Well database really is a bottleneck for me. I am currently trying to do rating stuff in agi using perl. What im doing is i lookup the rate of every dialed code for every call from the mysql database using the longest match technique. The longest match technique costs atleast 2-3 mysql queries for every call untill the dialed code is matched out of 14000 dialcodes. I dont know how to calculate the exact delay due to execution of agi, but on the asterisk cli whenever that agi executes, there is a visual delay of about half a sec to move from the agi extension to the next extension (can anybody tell me how to calculate the delay). Now im planning to use the manager api for constant connectivity to mysql and to enhance the longest match technique. Can anybody help me with this? Is it a good idea to ue manager api for lookingup the rate of the live call? On Sun, May 4, 2008 at 1:34 PM, Grey Man [EMAIL PROTECTED] wrote: If you've got anything but trivial AGI loads you should switch to FastAGI and put your business logic on a separate server to your Asterisk server. I use a deployment where a call could make up to 3 AGI requests per call before being put through (for things such as looking up accountcode, checking account credit, setting PSTN callerid). We monitor the time thw whole process takes and on average it's less than 100ms on an Asteisk server that peaks at 200 simultaneous calls (400 bridged) and 3 to 5 call set ups per second. The business logic processing the FastAGI calls is C# and .net which means Java would be able to handle it easily as well. The most likely bottleneck under high load will be your database. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Warning 2512
Hi all, I have been seeing a lot of the following warning messages on my asterisk cli. Can naybody tell why these messages are showing up. I am using only SIP to make calls from m asterisk. [Apr 17 04:52:24] WARNING[2512]: chan_sip.c:6480 determine_firstline_parts: Bad request protocol Bad event Also it will be great if anybody can tell where i can find the explanation of all the warnig codes and error codes of asterisk if there is any. -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Warning 2512
I just saw the sip debug and its showing that for every notify request, asterisk is sending a bad request response. here is the debug --- SIP read from 70.80.000.00:1031 --- NOTIFY sip:69.90.111.11:9060 SIP/2.0 Via: SIP/2.0/UDP 70.80.000.00:1032;branch=z9hG4bK-ade549bd From: Blake sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=90115683e082af23o0 To: sip:69.90.111.11 Call-ID: [EMAIL PROTECTED] CSeq: 7741 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA2102-5.2.3 Content-Length: 0 - --- (10 headers 0 lines) --- --- Transmitting (no NAT) to 70.80.000.00:1031 --- SIP/2.0 489 Bad event Via: SIP/2.0/UDP 70.80.000.00:1032;branch=z9hG4bK-ade549bd;received= 70.80.000.00 From: Blake sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=90115683e082af23o0 To: sip:69.90.111.11;tag=as3ef6a439 Call-ID: [EMAIL PROTECTED] CSeq: 7741 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Why is it doing so? On Thu, Apr 17, 2008 at 2:36 AM, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Rizwan Hisham [EMAIL PROTECTED] wrote: I have been seeing a lot of the following warning messages on my asterisk cli. Can naybody tell why these messages are showing up. I am using only SIP to make calls from m asterisk. [Apr 17 04:52:24] WARNING[2512]: chan_sip.c:6480 determine_firstline_parts: Bad request protocol Bad event Also it will be great if anybody can tell where i can find the explanation of all the warnig codes and error codes of asterisk if there is any. The [2512] is not a warning code. It is just the process ID of the Asterisk process or thread that generated the warning. The next part of your message (chan_sip.c:6480) shows the source file and line number where the error was generated. You can go to that point in the file to see what kind of checks it was making. You can also turn on SIP debugging at the Asterisk CLI prompt to see the packets sent to/from Asterisk. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] buying cards from pakistan
Hi all, i want to buy a pci or whatever card for asterisk to plug in my telephone line into it and use asterisk as a pbx. i have only one telephone line at home. can you recommend me a simple cheap card which i can buy in pakistan. I live in pakistan, and i dont know any dealers here who sell asterisk cards. if someone knows where to buy cards in pakistan, plz tell me about it. -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sample configuration files for ATAs
Hi all, I need some sample configuration files (in xml format) for some of my atas, spa-2102, 1001, 2002, 3000. If anybody can provide these, i'll be very glad. I have heard that some people can retrieve the configuration file from the ata. I have all the above mentioned ata's, so if you can tell me how to take out the conf files then it will also be very helpfull. -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Simple Question
Hi, Does anyone know the purpose of /n attached at the end of the dial command below Dial(Local/[EMAIL PROTECTED]/n Local/[EMAIL PROTECTED]/n) -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto-congest time for sip peers
Hi all, can anybody tell me how to make auto-congest time configurable(different) for every sip peer. I mean if i want to dial a local number then i should be able to set the autocongest time to 15000 mili seconds, but if i dial an international number then i should be able to set the auto-congest time to 30,000 mili seconds. Can it be done in the dialplan? or should i jump into the code? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The switch statement in extensions.conf
Hi all, i need to know how to use the switch statement in extensions.conf to throw calls on a secondary asterisk server. I have been trying all day to be able to make it work but its not working. It displays the following error: pbx_find_extension: No such switch 'master:[EMAIL PROTECTED]' I have read the help material on voip-info for connecting 2 asterisk servers and followed the steps given there but no use. Can anyone help me with this? Also need to know what the switch statement does internally. i mean does the server which switches the call to another server keeps any record of the call or it just transfers the call and then all of the responsibility of the call is handled on the other server? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID(num) not showing on cli
Hi, I just encountered a simple but strange problem. I am using 2 sip phones to call each other. Whenever i make a call, using softphone or ata, ali only shows the CallerID(name) and not the number. I have no idea why it does not show the number. I have tried various things but none have worked. The From header show the name and number which i set before dialing but on cli it shows only name: From: salman sip:[EMAIL PROTECTED]:5238;tag=as5100f7b2 Any one knows what should i do to solve this problem? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find out the IP of the calling party?
I dont know about IAX, but for SIP users you can use the function SIP_HEADER(headername) to get the information u need from the sip packets. for example you can use SIP_HEADER(From) which will give you the From header containing the IP address of the caller. You will have to apply regex on it to extract the ip. On Thu, Mar 13, 2008 at 8:47 AM, Gonzalo Servat [EMAIL PROTECTED] wrote: Hi All, I'm trying to achieve the following: - If sip/iax user logs in from home, they can dial internal extensions only (this is to avoid employees going wild on local/mobile calls from home) - If sip/iax user logs in from the office, they can call anyone they want. Since I have my users defined in an LDAP tree, I'd like to stick to one-account-per-user (each account is setup for both - IAX and SIP logins - to allow the employee to use IAX from home and SIP at work, or whatever combination they prefer). So, I thought I would simply look at the IP address of the originating call. If the SIP/IAX user has an IP address outside the local subnet - allow calls to extensions only. Else - allow all. I thought the best way of doing this would be using AGI with a Perl script. The only problem I'm having is determining the IP address of the originating call. I can't find any channel variable that gives me this info. The reason why I mentioned that I'd like to stick to one-account-per-user is that I know I could fix this simply by having 2 accounts per user (one that allows connections from the local subnet, and the other to login from outside and use different contexts for each), but it'd be much nicer to avoid having 2 accounts per user. If anyone has any suggestions on how to achieve the above, I'd love to read them! Thanks in advance. Gonzalo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as useragent registered using 2 accounts
Adding fromuser option in trunk declaration in AST1 has solved all problems though. On Wed, Feb 27, 2008 at 4:36 PM, Igor A. Goncharovsky [EMAIL PROTECTED] wrote: Rizwan Hisham wrote: I am having a strange problem. I am using my asterisk server AST1 to register with another asterisk server AST2 using 2 accounts (2 register commands in sip.conf). I have made 2 local users in AST1, and configured my dialplan in such a way that both local accounts on AST1 use different trunks to send the call to AST2 server. These 2 different trunks are for 2 accounts i have registered on AST1. (skiped) How can i make asterisk realize it? You must enable authentication of INVITE that AST1 send to AST2. Now you have no authentication of incoming INVITE and AST2 make decision about used account based only on IP address of caller peer. Changing insecure=port,invite to insecure=port should help. -- Best regards, Igor A. Goncharovsky ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as useragent registered using 2 accounts
Thanx for the tip. It has erased the problem i was having using authentication but another problem has arised. i am now able to call with only one user from AST1 to AST2. If i dial using the other user, my AST2 shows the following warning and responds with a 403 forbidden sip response: *WARNING[13520]: chan_sip.c:8117 check_auth: username mismatch, have adf, digest has abc* Any solutions to this problem? On Wed, Feb 27, 2008 at 4:36 PM, Igor A. Goncharovsky [EMAIL PROTECTED] wrote: Rizwan Hisham wrote: I am having a strange problem. I am using my asterisk server AST1 to register with another asterisk server AST2 using 2 accounts (2 register commands in sip.conf). I have made 2 local users in AST1, and configured my dialplan in such a way that both local accounts on AST1 use different trunks to send the call to AST2 server. These 2 different trunks are for 2 accounts i have registered on AST1. (skiped) How can i make asterisk realize it? You must enable authentication of INVITE that AST1 send to AST2. Now you have no authentication of incoming INVITE and AST2 make decision about used account based only on IP address of caller peer. Changing insecure=port,invite to insecure=port should help. -- Best regards, Igor A. Goncharovsky ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
