[asterisk-users] Cdr Logs modify Disposition on Unsuccessful call
Hi Team, I would like capture SS7 Error Code in CDRs, Specifically of outbound call from the asterisk. calls generated using .call file. In extension.conf extens gets excuted on successful call only , So that on h extension reason of hangup is captured. But i am not aware of any provision that capture on Unsuccessful call. please guide on this or suggest any patch. Thanks Vinod d -- _ -- 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] SS7 Disposition
Hi team, I am experience the same issue. Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 1 Mar 2012 15:32:41 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SS7 Disposition -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SS7 Disposition
Hi , Yes, I am using asterisk-java ami to originate call. Using LibSS7 Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Date: Thu, 1 Mar 2012 18:23:47 To: vdharash...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] SS7 Disposition Are you using AMI originate for these SS7 outbound calls? On Thu, Mar 1, 2012 at 6:15 PM, [Digital^Dude] ® millennium@gmail.comwrote: What versions on Asterisk and chan_ss7 are you using? On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote: Hi team, I am experience the same issue. Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 1 Mar 2012 15:32:41 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: [asterisk-users] SS7 Disposition -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SS7 Disposition
Hi , I am using asterisk 1.6.1, any idea patch for the same Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Date: Thu, 1 Mar 2012 19:58:13 To: vdharash...@gmail.com Cc: asterisk-userasterisk-users@lists.digium.com Subject: Re: [asterisk-users] SS7 Disposition I tried it on asterisk 1.8, and it worked fine. On Thu, Mar 1, 2012 at 6:39 PM, Vinod Dharashive vdharash...@gmail.comwrote: ** Hi , Yes, I am using asterisk-java ami to originate call. Using LibSS7 Thanks Vinod dharashive Sent from BlackBerry® on Airtel -- *From: * [Digital^Dude] ® millennium@gmail.com *Date: *Thu, 1 Mar 2012 18:23:47 +0500 *To: *vdharash...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Subject: *Re: [asterisk-users] SS7 Disposition Are you using AMI originate for these SS7 outbound calls? On Thu, Mar 1, 2012 at 6:15 PM, [Digital^Dude] ® millennium@gmail.com wrote: What versions on Asterisk and chan_ss7 are you using? On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote: Hi team, I am experience the same issue. Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 1 Mar 2012 15:32:41 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: [asterisk-users] SS7 Disposition -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to support Dialogic Cards
Hi Kevin, Is there any possibility of asterisk supported for dialogic cards in future. does digium has any plan for supporting it? Thanks Vinod Dharashive On Thu, Jan 19, 2012 at 9:02 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 01/19/2012 07:23 AM, virendra bhati wrote: Dialogic doesn't provide any soultion as open source. It provides hardware base cards for making outbond calls. And they used asterisk as backend for they card application. Dialogic cards are useful for both outbound *and* inbound calls. There have been various efforts to open-source drivers for some of them, but there are multiple families of Dialogic cards with differing hardware, so they don't all use the same drivers. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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-**usershttp://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
[asterisk-users] Asterisk to support Dialogic Cards
Hi Team, Is there any way that asterisk can support Dialogic card, i have done lot of search but could find any useful information. Thanks Vinod Dharashive -- _ -- 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] Chan_ss7 clustering config with single point
Hi team, Can any one share with me clustering configuration file SS7.conf for single pointcode with four slc. two different machine each host having 2 slc respectively. Thanks Vinod Dharashive Sent from BlackBerry® on Airtel -- _ -- 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 call time
Hi team, On event of no answer in CDR the starttime and endtime of call remains the same. Is there any way how can actually track call originate time and call end time. Thanks Vinod dharashive. Sent from BlackBerry® on Airtel -- _ -- 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] Sangoma Card with 16E1 SS7 signaling
Hi Team, i have been facing issues with sangoma card with 16 E1. used LibSS7 asterisk 1.6 with 8 E1 the links are stable , but moment i add another card of 8 E1 for to support 16 E1. link keeps fluctuating any idea why ? Please help Thanks Vinod Dharashive -- _ -- 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] Sangoma Card with 16E1 SS7 signaling
Hi Richard, The link is up on 16 th channel. My objective is to have 16 E1 to be configure on single machine with two 8 port sangoma card. Which is problem I am facing. Please let me know if you have any solution. Thanks Vinod dharashive --Original Message-- From: Richard Mudgett Sender: asterisk-users-boun...@lists.digium.com To: asterisk-user ReplyTo: asterisk-user Subject: Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling Sent: Oct 27, 2011 9:33 PM Hi Team, i have been facing issues with sangoma card with 16 E1. used LibSS7 asterisk 1.6 with 8 E1 the links are stable , but moment i add another card of 8 E1 for to support 16 E1. link keeps fluctuating any idea why ? Your 16th channel may be mismatched with the network. Timeslot 16 is usually used for signaling. channels = 1-15,17 Richard -- _ -- 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 Sent from BlackBerry® on Airtel -- _ -- 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] Sangoma Card with 16E1 SS7 signaling
Hi stefan, Do you have any configuration file for 16 E1 of system.conf and wanpipeX.conf. Can u please share with me. Thanks Vinod dharashive. --Original Message-- From: Stefan Schmidt To: vdharash...@gmail.com To: asterisk-user Subject: Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling Sent: Oct 27, 2011 11:23 PM Am 27.10.2011 19:29, schrieb Vinod Dharashive: Hi Richard, The link is up on 16 th channel. My objective is to have 16 E1 to be configure on single machine with two 8 port sangoma card. Which is problem I am facing. Please let me know if you have any solution. Thanks Vinod dharashive Hi vinod, i have never tried to put two A108 cards into one server but it sounds like an IRQ Problem to me. you should check all system relevant information (lsmod, lspci,dmesg) if you can see any kernel errors with this. it could also be a span identification problem that chan_dahdi recognize the second card also beginning with span 1 and not 9 or something like this. btw i dont think that you will be happy with asterisk 1.6 and handling 496 concurrent calls over ss7 best regards Stefan Sent from BlackBerry® on Airtel -- _ -- 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] Sangoma Card with 16E1 SS7 signaling
Hi Richard, Thanks for reply, how can I identify clocking issues on card. Manually calls goes successfully on both the card. After 10 min signaling links goes down. Thanks Vinod dharashive --Original Message-- From: Richard Mudgett To: vdharash...@gmail.com To: asterisk-user Subject: Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling Sent: Oct 27, 2011 11:34 PM Hi Team, i have been facing issues with sangoma card with 16 E1. used LibSS7 asterisk 1.6 with 8 E1 the links are stable , but moment i add another card of 8 E1 for to support 16 E1. link keeps fluctuating any idea why ? Your 16th channel may be mismatched with the network. Timeslot 16 is usually used for signaling. channels = 1-15,17 The link is up on 16 th channel. My objective is to have 16 E1 to be configure on single machine with two 8 port sangoma card. Which is problem I am facing. Please let me know if you have any solution. Sounds like it could be a clocking issue between the two cards then. Everything needs to use the same clock source. Richard Sent from BlackBerry® on Airtel -- _ -- 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] Sangoma Card with 16E1 SS7 signaling
Hi Patrick, I did that, sangoma does not support Libss7 stack and they mentioned that Libss7 does work right with sangoma card with 16 e1. Thanks Vinod Dharashive --Original Message-- From: Patrick Lists Sender: asterisk-users-boun...@lists.digium.com To: asterisk-user ReplyTo: asterisk-user Subject: Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling Sent: Oct 28, 2011 3:43 AM On 10/27/2011 08:57 PM, Vinod Dharashive wrote: Hi Richard, Thanks for reply, how can I identify clocking issues on card. Manually calls goes successfully on both the card. After 10 min signaling links goes down. Since you have a Sangoma card why don't you ask Sangoma support instead of asking Digium? Patrick -- _ -- 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 Sent from BlackBerry® on Airtel -- _ -- 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] (no subject)
Hi sam,
Have solved the problem with your advice. Call drop in 10 seconds without
disconnecting a-party call. Thank you very much.
[TB]
exten =_X.,1,Wait(${INCOMING_WAIT})
exten =_X.,2,Verbose(TB)
exten =_X.,3,Answer()
exten =_X.,4,Set(mainLoop=0)
;exten =_X.,5,Set(TIMEOUT(absolute)=5)
exten =_X.,5,Playback(/var/callagent/prompts/monitor/thanks)
exten = _X.,6,Dial(DAHDI/7/
09501032209,100,L(3[:1][:3000])g)
exten =_X.,7,Noop(${DIALEDTIME})
exten =_X.,8,Goto(TB,_X.,1)
exten =_X.,n,Hangup()
Cheers
Vinod Dharashive
Sent from BlackBerry® on Airtel
-Original Message-
From: Sam Govind govoi...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 7 Sep 2011 11:53:33
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] (no subject)
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[asterisk-users] (no subject)
Hi team,
I am trying to find solution to hangup b-party call after 1 min with out
disconnecting the call of a-party but following dial plan which is disconnect
both the calls.
Please suggest me the solution.
[TB]
exten = _X.,1,Wait(${INCOMING_WAIT})
exten =_X.,2,Verbose(TB)
exten =_X.,3,Answer()
exten = _X.,4,Set(mainLoop=0)
exten = _X.,5,Set(TIMEOUT(absolute)=10) ; set time in milliseconds
exten = _X.,6,Playback(/var/callagent/prompts/monitor/thanks)
exten = _X.,7,Dial(DAHDI/7/
09501032209,10,S(60))
exten = _X.,8,Noop(${DIALEDTIME})
exten =_X.,9,Goto(TB,_X.,1)
exten =_X.,n,Hangup()
Thanks
Vinod Dharashive
Sent from BlackBerry® on Airtel
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[asterisk-users] Accept the dtmf input in call patch
Hi team, Is it possible to capture dtmf input once call is patched between a-party and b-party? Also on dtmf input issue hangup request to b-party with out disconnecting A-party. How is this scenario implemented in dialplan? Thanks Vinod Dharashive Sent from BlackBerry® on Airtel -- _ -- 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] Dialplan required for recording
Hi team, Can any one help me to implement dialplan in which conversation between a-party and b-party (call patch) needs to be recorded. Thanks Vinod Dharashive Sent from BlackBerry® on Airtel -- _ -- 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
