[asterisk-users] AMI redirect from Queue to MeetMe
Hello List, I have scenario as follows, 1. A call comes to queue. 2. Available agent will answer the call. 3. BridgeEvent wil be generated in AMI with channel1 and channel2. 4. Parse channel1 and channel two from the event and redirect them to a meetme room, Dialplan, Exten = 1234,1,MeetMe(1234,1dq) But sometime it works and sometime one leg gets disconnected after redirection. Is it a bug to asterisk-1.6.2.13 ? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.comhttp://www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. Thank You. -- _ -- 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] Filtering on from caller id
In sip.conf you point to a context. For 101 en 102 you should point to a context that allows using the trunk while for the other numbers you doesn't grand this privilege . Erik Verstuurd vanaf mijn iPad Op 24 mrt. 2011 om 16:58 heeft Peter den Hartog peterdenhar...@gmail.com het volgende geschreven: Hi, I would like to use the from caller id, to allow calls yes or no. 101, and 111 should be allowed to use the Trunk, the rest of the phones are not. Is this even possible? So if the from caller id is 101 or 111, then allow the call, otherwise hangup. Thanks, Peter -- _ -- 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
[asterisk-users] problems with blind transfer on GXP-2000 - Multi tenant asterisk !!
Hello Users, We have Thirdlane Multi tenant PBX system in production. Asterisk version is 1.6.2.15. Attendant transfer is working, but blind transfer is not working with Grandstream (gxp-2000) phone. We have read from google that it is a bug in Asterisk 1.6.2.15. We saw the below links: http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fwww.freepbx.o rg%252Fforum%252Ffreepbx%252Fusers%252Ftransfer-bug-on-asterisk-1-4-38-1-6-2 -15-and-1-8-0-1 http://www.freepbx.org/forum/freepbx/users/transfer-bug-on-asterisk-1-4-38-1 -6-2-15-and-1-8-0-1 http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fasteriskfaqs. org%252F2010%252F10%252F21%252Fasterisk-users%252Fblind-transfer-failed-sip- refer-method.html http://asteriskfaqs.org/2010/10/21/asterisk-users/blind-transfer-failed-sip- refer-method.html http://www.odesk.com/leaving_odesk.php?ref=https%253A%252F%252Fissues.aster isk.org%252Fview.php%253Fid%253D18185 https://issues.asterisk.org/view.php?id=18185 http://www.odesk.com/leaving_odesk.php?ref=https%253A%252F%252Fissues.aster isk.org%252Fview.php%253Fid%253D18185%2523c128539 https://issues.asterisk.org/view.php?id=18185#c128539 How to do this: We make a call from number to DDI, assigned to a Tenant in PBX. Incoming call is routed to an extension 100. We press transfer button and press the extensions. Call is dropped immediately. We couldn't see any error in the SIP log. I have attached the sip log, where call is made to ddi 016700202 and 100-solitaire rang. blindtransfer was made to 101-solitaire. If anyone is interested to fix this for us, it will be good. My yahoo id is sreedha...@yahoo.com Myskypeid is kvss2010 Note: I can provide more information upon request. Regs Sreedhar. -- _ -- 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] why does core show channels on 1.8 not show the channel
Maybe this helps: https://issues.asterisk.org/view.php?id=18603 Arjan -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Jerry Geis Verzonden: 20-03-2011 21:24 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [asterisk-users] why does core show channels on 1.8 not show the channel When I do core show channels on 1.8 it gives me something like: Channel Location State Application(Data) DAHDI/i1/3175551212- s@default:10 Up BackGround(SM_ATTENDANT) 1 active channel 1 active call 188 calls processed No active MeetMe conferences. What channel is i1?? It used to show me DAHDI/18/3175551212 . How do I relate i1 to 18 which is the real channel. 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 -- _ -- 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] Queue(): how to Perform operations at the time of call sent to Queue member but not answered.
Hi Group, In Queue application, we have AGI,macro and gosub parameters that allow us to perform some operations when Queue member gets connected with caller. But it seems that right now there is no such mechanism (except CEL,AMI) for situation where we want some operations to be performed when call is sent to Queue Member but not answered yet (i.e. Queue Member interface is in ringing state). I know monitoring Channel events we can do this, but I wanted something in dialplan itself to get it done. Probably I will not be the only person asking for this future in Asterisk. PLease have your thoughts on this. Thanking you. --AM -- _ -- 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 help: hang up incoming call and call the number back
Hi, I'm trying to setup Asterisk so that: 1. I call a specific number that goes to a defined extension from my phone (an external line). 2. Asterisk notes my phone number (the CLID) and hangs up without picking up the call. 3. Asterisk initiates a call to my phone and prompts me for a passkey. 4. Asterisk validates the passkey and lets me enter another number (say FOO). 5. Asterisk dials FOO on my behalf and lets me talk to FOO. I'm currently using extension h for handling the post-hangup processing; however this seems to involve a lot of validation (since other calls are also terminating into the same context and, eventually, getting hungup) and has issues passing variables between extensions. Is there a better way of handling the post-hangup processing? Regards, -- Raj -- _ -- 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] Dialplan help: hang up incoming call and call the number back
On Mon, Mar 28, 2011 at 05:14:50PM +0530, Raj Mathur wrote: Is there a better way of handling the post-hangup processing? Callfiles? -- _ -- 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] Checking status of a cell phone
On Sat, 26 Mar 2011 14:58:30 +0100, magnu...@inputinterior.se wrote: Celluar Network - E1 - Avaya - OOH323 - Asterisk Thanks for the tip. So here's how it works: 1. The web app calls a script that uses AMI + Originate to send a call to the Avaya PBX 2. Avaya is able to check that a number (cellphone in this case) is busy and calls a different number in Asterisk to indicate the status through a value in the DB 3. The web script reads the value of DS/0733025975 and displays the status -- _ -- 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] Checking status of a cell phone
Its not the Avaya that makes the call back, it is mobile. -Ursprungligt meddelande- From: Gilles Sent: Monday, March 28, 2011 1:57 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Checking status of a cell phone On Sat, 26 Mar 2011 14:58:30 +0100, magnu...@inputinterior.se wrote: Celluar Network - E1 - Avaya - OOH323 - Asterisk Thanks for the tip. So here's how it works: 1. The web app calls a script that uses AMI + Originate to send a call to the Avaya PBX 2. Avaya is able to check that a number (cellphone in this case) is busy and calls a different number in Asterisk to indicate the status through a value in the DB 3. The web script reads the value of DS/0733025975 and displays the status -- _ -- 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
[asterisk-users] asterisk and fail2ban
Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in maxretry in jail.conf For example, I get an email saying: The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts against ASTERISK. when maxretry = 5 in jail.conf Perhaps someone else is experiencing this or has resolved it, thank you in advance for your time. -- _ -- 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 and fail2ban
On Mon, Mar 28, 2011 at 9:20 AM, vip killa vipki...@gmail.com wrote: Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in maxretry in jail.conf For example, I get an email saying: The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts against ASTERISK. when maxretry = 5 in jail.conf Perhaps someone else is experiencing this or has resolved it, thank you in advance for your time. If you fixed the logging issue discussed here http://www.fail2ban.org/wiki/index.php/Asterisk then I would assume your logging has problems. -- ~~~ Andrew lathama Latham lath...@gmail.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] Variable. AMI and dialplan
Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what. From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value: 5\r\n\r\n From dialplan i can “access” the variable “x” and see the value “5” From dialplan i modify “x” to “8”. But from AMI i still se “x” as “5” not “8”. /Magnus-- _ -- 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] DTMF input while waiting in queue...
Hey all!
I'm trying to figure out how to have a queue accept an inbound caller's key
press to action on. At first I'm just trying to implement a Press 1 to
leave a voice mail announced and at any time in the queue, the user can
press 1 and go to the queue's voicemail. Later I'd like to have it accept
Press 1 if this is an x issue, press 2 if this a y problem and I'll have
UserEvent's generated for the press. This is the dial plan I have now.
exten = 1820,1,Macro(user-callerid,)
exten = 1820,n,Answer
exten = 1820,n,ExecIf($[${QUEUEWAIT} = ]?Set(__QUEUEWAIT=${EPOCH}))
exten = 1820,n,Set(__BLKVM_OVERRIDE=BLKVM/${EXTEN}/${CHANNEL})
exten = 1820,n,Set(__BLKVM_BASE=${EXTEN})
exten = 1820,n,Set(DB(${BLKVM_OVERRIDE})=TRUE)
exten = 1820,n,UserEvent(QueueAlert,CallerID: ${CALLERID(number)},Queue:
${EXTEN},UniqueID: ${UNIQUEID},Channel: ${CHANNEL})
exten = 1820,n,ExecIf($[${REGEX((M[(]auto-blkvm[)]) ${DIAL_OPTIONS})}
!= 1]?Set(_DIAL_OPTIONS=${DIAL_OPTIONS}M(auto-blkvm)U(ackcall^s^1)))
exten = 1820,n,Set(__NODEST=${EXTEN})
exten = 1820,n,GotoIf($[foo${RGPREFIX} = foo]?REPCID)
exten = 1820,n,GotoIf($[${RGPREFIX} !=
${CALLERID(name):0:${LEN(${RGPREFIX})}}]?REPCID)
exten = 1820,n,Noop(Current RGPREFIX is ${RGPREFIX}stripping from
Caller ID)
exten = 1820,n,Set(CALLERID(name)=${CALLERID(name):${LEN(${RGPREFIX})}})
exten = 1820,n,Set(_RGPREFIX=)
exten = 1820,n(REPCID),Noop(CALLERID(name) is ${CALLERID(name)})
exten = 1820,n,Set(_RGPREFIX=IT HD:)
exten = 1820,n,Set(CALLERID(name)=${RGPREFIX}${CALLERID(name)})
exten =
1820,n,Set(MONITOR_FILENAME=/var/spool/asterisk/monitor/q${EXTEN}-${STRFTIME
(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID})
exten = 1820,n,Set(__CWIGNORE=TRUE)
exten = 1820,n,Queue(1820,t,,)
exten = 1820,n,Noop(Deleting: ${BLKVM_OVERRIDE}
${DB_DELETE(${BLKVM_OVERRIDE})})
exten = 1820,n,Set(__NODEST=)
exten = 1820,n,Set(__CWIGNORE=)
exten = 1820,n,Goto()
exten = 1820*,1,Macro(agent-add,1820,1739)
exten = 1820**,1,Macro(agent-del,1820)
exten = *451820,1,Set(QUEUENO=1820)
exten = *451820,n,Goto(app-queue-toggle,s,start)
exten = h,1,Macro(hangupcall,)
Thanks in advance everyone!
Louis
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[asterisk-users] DAHDI, IAX2 and SIP considerations for Early-Media / Alerting
Hi, Short version: Is it possible or even legal to convert an IAX2 PROGRESS/EARLY-MEDIA indication into a DAHDI/q.931 ALERTING signal when your ISDN provider does not pass early media on receipt of an PROGRESS(8) indication? Long version: I have an Asterisk 1.6.2.18-rc1 based system with a DAHDI trunk (UK E1 line), also, the system has IAX2 trunks, and several SIP handsets. All 3 protocols (q.931/IAX2/SIP) have a mechanism to indicate either ALERTING/RINGING, or to specify PROGRESS/EARLY-MEDIA. Based on this you'd think call setup would all work happily all of the time :) What happens based on the call direction is as follows: SIP - DAHDIISDN returns ALERTING, SIP uses 180 Ringing, all OK SIP - IAX2 IAX2 returns PROGRESS, SIP uses 183 Progress, early audio works OK IAX2 - DAHDI ISDN returns ALERTING, IAX2 uses RINGING, all OK IAX2 - SIP SIP returns 180 ringing, IAX2 uses RINGING, all OK DAHDI - SIPSIP returns 180 ringing, ISDN uses ALERTING, all OK DAHDI - IAX2 IAX2 returns PROGRESS, ISDN uses PROGRESS(8), but the caller hears no ringing. I believe that my issue is that my UK ISDN provider does not accept early media, and will simply send silence instead of using the provided early audio stream. DAHDI is configured with: priindication=outofband The IAX2 trunk provider is using early-media to send the ringing tone, and as above, this mostly seems to work okay. The exception is when the call is bridged to ISDN, where I believe the ISDN provider does not pass on early media. I checked the IAX2 RFCs 5456/5457, but cannot find a definition of how RINGING/PROGRESS is meant to work. Is my IAX2 trunk provider doing something wring by not also sending RINGING? Is there a workaround that converts either IAX2 PROGRESS into RINGING, or allows DAHDI to send ALERTING if it receives an early media indication? I suspect the code to do the latter would be reasonably simple, but would appreciate pointers for any badness that it may cause. Thanks in advance for any suggestions. Regards, Steve -- _ -- 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 MYSQL missing field data
Thanks Tilghman for your response. I have the following in my cdr_mysql.conf I put it in sometime yesterday and did not have it till then. However, it did not make any difference. [columns] static value = column alias cdrvar = column alias start = calldate alias callerid = clid alias src = src alias dst = dst alias dcontext = dcontext alias channel = channel alias dstchannel = dstchannel alias lastapp = lastapp alias lastdata = lastdata alias duration = duration alias billsec = billsec alias disposition = disposition alias amaflags = amaflags alias accountcode = accountcode alias userfield = userfield alias uniqueid = uniqueid Thanks again Eric -- Eric W. Davenport Cert-In Software Systems, Inc. P.O. Box 346 Bakersville, NC 28705 800-873-0110 ewdavenp...@certin.com www.certin.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] Variable. AMI and dialplan
On 3/28/2011 7:41 AM, magnu...@inputinterior.se wrote: Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what. From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value: 5\r\n\r\n From dialplan i can “access” the variable “x” and see the value “5” From dialplan i modify “x” to “8”. But from AMI i still se “x” as “5” not “8”. /Magnus Maybe you need to perform a GetVar to read the new value of that channel variable -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions 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
[asterisk-users] Channel status with AMI originate calls
Hi, is there a way to know if originate call channel ended the call *before* connecting to context/extension/priority? DIALSTATUS is empty, HANGUPCAUSE is always 16, nothing in SIP Headers nor in AST_CONTROL_FRAME_[HANGUP|ANSWER] Asterisk is 1.6.2.16 Thanks for any hint -- Daniel -- _ -- 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] Variable. AMI and dialplan
I did use Action: Getvar when i read it back in AMI. On 3/28/2011 7:41 AM, magnu...@inputinterior.se wrote: Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what. From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value: 5\r\n\r\n From dialplan i can “access” the variable “x” and see the value “5” From dialplan i modify “x” to “8”. But from AMI i still se “x” as “5” not “8”. /Magnus Maybe you need to perform a GetVar to read the new value of that channel variable -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions 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
Re: [asterisk-users] asterisk and fail2ban
Yes I followed directions on that page Running Asterisk 1.6.1.22, anybody else experiencing this? On Mon, Mar 28, 2011 at 8:32 AM, Andrew Latham lath...@gmail.com wrote: On Mon, Mar 28, 2011 at 9:20 AM, vip killa vipki...@gmail.com wrote: Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in maxretry in jail.conf For example, I get an email saying: The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts against ASTERISK. when maxretry = 5 in jail.conf Perhaps someone else is experiencing this or has resolved it, thank you in advance for your time. If you fixed the logging issue discussed here http://www.fail2ban.org/wiki/index.php/Asterisk then I would assume your logging has problems. -- ~~~ Andrew lathama Latham lath...@gmail.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 -- _ -- 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] Variable. AMI and dialplan
Don't know then, that's all I've got far ya today mate, sorry On 3/28/2011 8:18 AM, magnu...@inputinterior.se wrote: I did use Action: Getvar when i read it back in AMI. On 3/28/2011 7:41 AM, magnu...@inputinterior.se wrote: Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what. From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value: 5\r\n\r\n From dialplan i can “access” the variable “x” and see the value “5” From dialplan i modify “x” to “8”. But from AMI i still se “x” as “5” not “8”. /Magnus Maybe you need to perform a GetVar to read the new value of that channel variable -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions 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
[asterisk-users] Asterisk SS7 error
Hi Asterisks Team, I am getting the error below after getting a connection to a telco using ss7. Anyone know how to solve it? The link keeps coming up and down every 30 seconds. Resetting CIC 3 [Mar 28 15:16:28] WARNING[20434]: chan_dahdi.c:11913 ss7_linkset: RSC on unconfigured CIC 3 Received out of sequence MSU w/ fsn of 119, lastfsnacked = 116, requesting retransmission MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. Resetting CIC 4 [Mar 28 15:16:28] WARNING[20434]: chan_dahdi.c:11913 ss7_linkset: RSC on unconfigured CIC 4 Received out of sequence MSU w/ fsn of 119, lastfsnacked = 117, requesting retransmission MSU received, though still waiting for retransmission start. Dropping. MSU received, though still waiting for retransmission start. Dropping. Thanks for your help in advance. FYI am using Asterisks 1.6 Peter. -- _ -- 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 and fail2ban
On 28 Mar 2011, at 14:19, vip killa wrote: Yes I followed directions on that page Running Asterisk 1.6.1.22, anybody else experiencing this? How often does fail2ban check the logs? It can only block that often, so if more attempts happen in that time period it can't do anything until it knows. 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
Re: [asterisk-users] Variable. AMI and dialplan
this may be related with: https://issues.asterisk.org/view.php?id=14662 El 28/03/2011 10:20, Sherwood McGowan escribió: Don't know then, that's all I've got far ya today mate, sorry On 3/28/2011 8:18 AM, magnu...@inputinterior.se wrote: I did use Action: Getvar when i read it back in AMI. On 3/28/2011 7:41 AM, magnu...@inputinterior.se wrote: Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what. From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value: 5\r\n\r\n From dialplan i can “access” the variable “x” and see the value “5” From dialplan i modify “x” to “8”. But from AMI i still se “x” as “5” not “8”. /Magnus Maybe you need to perform a GetVar to read the new value of that channel variable -- _ -- 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] Sox and bad quality when converting to 8 kHz
On Thu, Mar 24, 2011 at 4:58 PM, Thomas Winter thowin...@googlemail.com wrote: Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help I have had good results with this: sox in.wav -r 8000 -c 1 out.wav highpass 500 lowpass 4000 resample -ql Play around with the high and low pass numbers because they might need to be changed depending on the properties of your recordings. -M -- _ -- 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 and fail2ban
fail2ban checks the logs every second. Does asterisk buffer log output? On Mon, Mar 28, 2011 at 9:27 AM, Steven Howes steve-li...@geekinter.netwrote: On 28 Mar 2011, at 14:19, vip killa wrote: Yes I followed directions on that page Running Asterisk 1.6.1.22, anybody else experiencing this? How often does fail2ban check the logs? It can only block that often, so if more attempts happen in that time period it can't do anything until it knows. 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 -- _ -- 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 redirect from Queue to MeetMe
I would be surprised that you did not always hang up the second channel you are
redirecting. Once you transfer one leg there is nothing connected to the second
leg so it goes away, I would think.
What we do is remember the agent number, transfer the caller, and then setup a
call to the agent and meetme room.
More or less like:
Action: Redirect
Channel: SIP/GXP280_18-0001
Exten: do_meetme601MyID
Context: cfmc_cdi_private
Priority: 1
ActionID: MeetMe
Async: true
Action: Originate
Channel: Agent/1001
Exten: do_meetme601MyID2
Context: cfmc_cdi_private
Priority: 1
ActionID: DirectMeet
Async: true
exten = _do_meetme.,1,UserEvent(BeforeMeetMe,Info:${EXTEN:9} ${UNIQUEID}
${CHANNEL})
exten = _do_meetme.,n,Answer()
exten = _do_meetme.,n,Set(CfMC_RoomToUse=${EXTEN:9:3})
exten = _do_meetme.,n,Set(CfMC_CurrentID=${EXTEN:12})
exten = _do_meetme.,n,Set(MEETME_MOH_CLASS=meetme-music)
exten = _do_meetme.,n,MeetMe(${CfMC_RoomToUse},CMpqx1)
exten = _do_meetme.,n,UserEvent(AfterMeetMe,ActionID:${CfMC_CurrentID}
Room:${CfMC_RoomToUse} ${UNIQUEID} ${CHANNEL})
exten = _do_meetme.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 28, 2011, at 1:23 AM, Deka, Rajib IN MAA SL wrote:
Hello List,
I have scenario as follows,
A call comes to queue.
Available agent will answer the call.
BridgeEvent wil be generated in AMI with channel1 and channel2.
Parse channel1 and channel two from the event and redirect them to a meetme
room,
Dialplan,
Exten = 1234,1,MeetMe(1234,1dq)
But sometime it works and sometime one leg gets disconnected after
redirection. Is it a bug to asterisk-1.6.2.13 ?
Regards,
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com
Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com
Important notice: This e-mail and any attachment there to contains corporate
proprietary information. If you have received it by mistake, please notify us
immediately by reply e-mail and delete this e-mail and its attachments from
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Thank You.
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Re: [asterisk-users] DTMF input while waiting in queue...
On 3/28/2011 7:54 AM, Louis Carreiro wrote: Hey all! I’m trying to figure out how to have a queue accept an inbound caller’s key press to action on. At first I’m just trying to implement a “Press 1 to leave a voice mail” announced and at any time in the queue, the user can press 1 and go to the queue’s voicemail. Later I’d like to have it accept “Press 1 if this is an x issue, press 2 if this a y problem” and I’ll have UserEvent’s generated for the press. *snip* In your queues.conf, in the definition for 1820, add the following: context=queue1820-exit Then, in your dialplan create a new context: [queue1820-exit] exten = 1,1,Noop(Caller Pressed 1 to leave a voicemail) exten = 1,n,Voicemail(voicemailbox,theoptionsyouwant) exten = 1,n,Hangup That should get you started...Read about the context configuration option here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf Cheers! -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions 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
Re: [asterisk-users] AMI redirect from Queue to MeetMe
On 3/28/2011 10:02 AM, Jim Dickenson wrote:
I would be surprised that you did not always hang up the second
channel you are redirecting. Once you transfer one leg there is
nothing connected to the second leg so it goes away, I would think.
What we do is remember the agent number, transfer the caller, and then
setup a call to the agent and meetme room.
More or less like:
Action: Redirect
Channel: SIP/GXP280_18-0001
Exten: do_meetme601MyID
Context: cfmc_cdi_private
Priority: 1
ActionID: MeetMe
Async: true
Action: Originate
Channel: Agent/1001
Exten: do_meetme601MyID2
Context: cfmc_cdi_private
Priority: 1
ActionID: DirectMeet
Async: true
exten = _do_meetme.,1,UserEvent(BeforeMeetMe,Info:${EXTEN:9}
${UNIQUEID} ${CHANNEL})
exten = _do_meetme.,n,Answer()
exten = _do_meetme.,n,Set(CfMC_RoomToUse=${EXTEN:9:3})
exten = _do_meetme.,n,Set(CfMC_CurrentID=${EXTEN:12})
exten = _do_meetme.,n,Set(MEETME_MOH_CLASS=meetme-music)
exten = _do_meetme.,n,MeetMe(${CfMC_RoomToUse},CMpqx1)
exten =
_do_meetme.,n,UserEvent(AfterMeetMe,ActionID:${CfMC_CurrentID}
Room:${CfMC_RoomToUse} ${UNIQUEID} ${CHANNEL})
exten = _do_meetme.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 28, 2011, at 1:23 AM, Deka, Rajib IN MAA SL wrote:
Hello List,
I have scenario as follows,
1. A call comes to queue.
2. Available agent will answer the call.
3. BridgeEvent wil be generated in AMI with channel1 and channel2.
4. Parse channel1 and channel two from the event and redirect them
to a meetme room,
Dialplan,
Exten = 1234,1,MeetMe(1234,1dq)
But sometime it works and sometime one leg gets disconnected after
redirection. Is it a bug to asterisk-1.6.2.13 ?
Regards,
*Rajib Deka*
SIEMENS Ltd.
RobertV Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com http://www.siemens.com/
Mob: +91-9176780669| E-Mail: rajib.d...@siemens.com
mailto:rajib.d...@siemens.com
Important notice: This e-mail and any attachment there to contains
corporate proprietary information. If you have received it by
mistake, please notify us immediately by reply e-mail and delete this
e-mail and its attachments from your system.
Thank You.
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You could also use the Extra options (Channel, context, extension,
priority) to transfer BOTH legs
--
Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] Variable. AMI and dialplan
Could be, do u think its a bug or do u think I am doing totally wrong? I can easily reproduce it if any needs more info. -Ursprungligt meddelande- From: Sebastian Sent: Monday, March 28, 2011 3:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Variable. AMI and dialplan this may be related with: https://issues.asterisk.org/view.php?id=14662 El 28/03/2011 10:20, Sherwood McGowan escribió: Don't know then, that's all I've got far ya today mate, sorry On 3/28/2011 8:18 AM, magnu...@inputinterior.se wrote: I did use Action: Getvar when i read it back in AMI. On 3/28/2011 7:41 AM, magnu...@inputinterior.se wrote: Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what. From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value: 5\r\n\r\n From dialplan i can “access” the variable “x” and see the value “5” From dialplan i modify “x” to “8”. But from AMI i still se “x” as “5” not “8”. /Magnus Maybe you need to perform a GetVar to read the new value of that channel variable -- _ -- 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
[asterisk-users] s extension not working
Hey Guys! I have asterisk 1.8.x and somehow my 's' extension not picking up any incoming calls.. Not working [from-pstn] exten = s,1,Answer() same = n,Playback(hello-world) same = n,Hangup() Working... [from-pstn] exten = _,1,Answer() same = n,Playback(hello-world) same = n,Hangup() -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
Re: [asterisk-users] s extension not working
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, March 28, 2011 11:04 AM To: asterisk-users Subject: [asterisk-users] s extension not working Hey Guys! I have asterisk 1.8.x and somehow my 's' extension not picking up any incoming calls.. Not working [from-pstn] exten = s,1,Answer() same = n,Playback(hello-world) same = n,Hangup() Working... [from-pstn] exten = _,1,Answer() same = n,Playback(hello-world) same = n,Hangup() -S Ok Satish. I assume sip.conf or dahdi.conf has a context of from-pstn. The key to actually solving this will be for you to give us say 10 lines of CLI output. -- _ -- 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] special control 16
Hi What is special control 16? I am getting this error quite often -- special control 16, then for some reason it puts on hold and then logs is full of Asked to transmit frame type ulaw, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) Both peer and trunk have same codec priority (disallow=all then allow=alaw then alllow=ulaw) Any ideas how to fix this ? -- SIP/zzz-01cd is making progress passing it to SIP/0010777-01cc -- SIP/0010777-01cc requested special control 16, passing it to SIP/zzz-01cd -- Started music on hold, class 'default', on SIP/zzz-01cd -- SIP/0010777-01cc requested special control 20, passing it to SIP/zzz-01cd -- SIP/0010777-01cc requested special control 16, passing it to SIP/zzz-01cd -- Stopped music on hold on SIP/zzz-01cd -- Started music on hold, class 'default', on SIP/zzz-01cd -- SIP/0010777-01cc requested special control 20, passing it to SIP/zzz-01cd -- Remote UNIX connection -- Remote UNIX connection disconnected -- SIP/zzz-01cd answered SIP/0010777-01cc [2011-03-28 18:22:27] DEBUG[22502]: channel.c:6092 ast_set_owners_and_peers: setting peeraccount to 123456 for SIP/0010777-01cc from data on channel SIP/zzz-01cd -- Locally bridging SIP/0010777-01cc and SIP/zzz-01cd [2011-03-28 18:22:27] WARNING[22502]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) [2011-03-28 18:22:27] WARNING[22502]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) [2011-03-28 18:22:27] WARNING[22502]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) [2011-03-28 18:22:27] WARNING[22502]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) -- Nick Ustinov -- _ -- 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] s extension not working
If i use 's' then i got following error. This scenario is back to back asterisk connected on PRI line (T1). for testing purpose i calling from one asterisk to other and i want to land call on 's' extension. shirley*CLI -- Extension '7527' in context 'from-pstn' from '7623' does not exist. Rejecting call on channel 0/1, span 1 If i use _XXX then it working with following output. shirley*CLI -- Accepting call from '7623' to '7527' on channel 0/1, span 1 -- Executing [7527@from-pstn:1] Answer(DAHDI/i1/7623-10, ) in new stack -- Executing [7527@from-pstn:2] Playback(DAHDI/i1/7623-10, hello-world) in new stack -- DAHDI/i1/7623-10 Playing 'hello-world.ulaw' (language 'en') -- Executing [7527@from-pstn:3] Hangup(DAHDI/i1/7623-10, ) in new stack == Spawn extension (from-pstn, 7527, 3) exited non-zero on 'DAHDI/i1/7623-10' -- Hungup 'DAHDI/i1/7623-10' From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 28 Mar 2011 11:08:57 -0500 Subject: Re: [asterisk-users] s extension not working From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, March 28, 2011 11:04 AM To: asterisk-users Subject: [asterisk-users] s extension not working Hey Guys! I have asterisk 1.8.x and somehow my 's' extension not picking up any incoming calls.. Not working [from-pstn] exten = s,1,Answer() same = n,Playback(hello-world) same = n,Hangup() Working... [from-pstn] exten = _,1,Answer() same = n,Playback(hello-world) same = n,Hangup() -S Ok Satish. I assume sip.conf or dahdi.conf has a context of from-pstn. The key to actually solving this will be for you to give us say 10 lines of CLI output. -- _ -- 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
[asterisk-users] Is History-Info (RFC4244) supported ?
Hi, Googling, I came across this document http://www.cytek.biz/roller/designbox/entry/asterisk_diversion_and_history_info which says History-Info header is supported in asterisk. Unfortunately, some details are missing (aka asterisk version). Reading latest 1.8 changelog does say much about this. As someone met any success implementing this feature (with Polycom 3.1 enabled phones, for instance) ? Regards -- _ -- 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] s extension not working
Uhm That's because you're being passed 7527 as the extension, so it won't match s On 3/28/2011 11:38 AM, satish patel wrote: If i use 's' then i got following error. This scenario is back to back asterisk connected on PRI line (T1). for testing purpose i calling from one asterisk to other and i want to land call on 's' extension. shirley*CLI -- Extension '7527' in context 'from-pstn' from '7623' does not exist. Rejecting call on channel 0/1, span 1 If i use _XXX then it working with following output. shirley*CLI -- Accepting call from '7623' to '7527' on channel 0/1, span 1 -- Executing [7527@from-pstn:1] Answer(DAHDI/i1/7623-10, ) in new stack -- Executing [7527@from-pstn:2] Playback(DAHDI/i1/7623-10, hello-world) in new stack -- DAHDI/i1/7623-10 Playing 'hello-world.ulaw' (language 'en') -- Executing [7527@from-pstn:3] Hangup(DAHDI/i1/7623-10, ) in new stack == Spawn extension (from-pstn, 7527, 3) exited non-zero on 'DAHDI/i1/7623-10' -- Hungup 'DAHDI/i1/7623-10' From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 28 Mar 2011 11:08:57 -0500 Subject: Re: [asterisk-users] s extension not working *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *satish patel *Sent:* Monday, March 28, 2011 11:04 AM *To:* asterisk-users *Subject:* [asterisk-users] s extension not working Hey Guys! I have asterisk 1.8.x and somehow my 's' extension not picking up any incoming calls.. Not working [from-pstn] exten = s,1,Answer() same = n,Playback(hello-world) same = n,Hangup() Working... [from-pstn] exten = _,1,Answer() same = n,Playback(hello-world) same = n,Hangup() -S Ok Satish. I assume sip.conf or dahdi.conf has a context of from-pstn. The key to actually solving this will be for you to give us say 10 lines of CLI output. -- _ -- 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 -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions 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
Re: [asterisk-users] s extension not working
@Sherwood, I was also thinking about that But then how 's' extension match any unknown number ? Like when call coming from PSTN then how IVR picked up...? -Satish Date: Mon, 28 Mar 2011 12:58:28 -0500 From: sherwood.mcgo...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] s extension not working Uhm That's because you're being passed 7527 as the extension, so it won't match s On 3/28/2011 11:38 AM, satish patel wrote: If i use 's' then i got following error. This scenario is back to back asterisk connected on PRI line (T1). for testing purpose i calling from one asterisk to other and i want to land call on 's' extension. shirley*CLI -- Extension '7527' in context 'from-pstn' from '7623' does not exist. Rejecting call on channel 0/1, span 1 If i use _XXX then it working with following output. shirley*CLI -- Accepting call from '7623' to '7527' on channel 0/1, span 1 -- Executing [7527@from-pstn:1] Answer(DAHDI/i1/7623-10, ) in new stack -- Executing [7527@from-pstn:2] Playback(DAHDI/i1/7623-10, hello-world) in new stack -- DAHDI/i1/7623-10 Playing 'hello-world.ulaw' (language 'en') -- Executing [7527@from-pstn:3] Hangup(DAHDI/i1/7623-10, ) in new stack == Spawn extension (from-pstn, 7527, 3) exited non-zero on 'DAHDI/i1/7623-10' -- Hungup 'DAHDI/i1/7623-10' From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 28 Mar 2011 11:08:57 -0500 Subject: Re: [asterisk-users] s extension not working *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *satish patel *Sent:* Monday, March 28, 2011 11:04 AM *To:* asterisk-users *Subject:* [asterisk-users] s extension not working Hey Guys! I have asterisk 1.8.x and somehow my 's' extension not picking up any incoming calls.. Not working [from-pstn] exten = s,1,Answer() same = n,Playback(hello-world) same = n,Hangup() Working... [from-pstn] exten = _,1,Answer() same = n,Playback(hello-world) same = n,Hangup() -S Ok Satish. I assume sip.conf or dahdi.conf has a context of from-pstn. The key to actually solving this will be for you to give us say 10 lines of CLI output. -- _ -- 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 -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions 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 -- _ -- 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] s extension not working
On 3/28/2011 1:33 PM, satish patel wrote: @Sherwood, I was also thinking about that But then how 's' extension match any unknown number ? Like when call coming from PSTN then how IVR picked up...? -Satish The 's' extension does not match anything other than 's'. If your sip registrations are configured without a trailing /DIDNUMBER, it gets sent to the 's' extension on your default context. However, if you want to match *any* number, you'd want '_X.', which matches any number. -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions 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
Re: [asterisk-users] CDR MYSQL missing field data
On Monday 28 March 2011 07:57:10 Eric W. Davenport wrote: Thanks Tilghman for your response. I have the following in my cdr_mysql.conf I put it in sometime yesterday and did not have it till then. However, it did not make any difference. Did you reload after making the change to the config file? [columns] static value = column alias cdrvar = column These are bogus and should never have been uncommented. alias start = calldate alias callerid = clid These are fine. alias src = src alias dst = dst alias dcontext = dcontext alias channel = channel alias dstchannel = dstchannel alias lastapp = lastapp alias lastdata = lastdata alias duration = duration alias billsec = billsec alias disposition = disposition alias amaflags = amaflags alias accountcode = accountcode alias userfield = userfield alias uniqueid = uniqueid There is no reason to have any of these uncommented, unless the column specified after the arrow is different from the field specified before. -- Tilghman -- _ -- 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 1.8 Packages for Debian and Ubuntu
- Original Message - Thanks for providing these - can you just clarify your policy on the following: - file locations - same layout as the regular Debian packages? Yes, same layout. - upgrade policy - is it intended that someone who has Debian 6 with the existing Asterisk 1.6 packages (from Debian's maintainer) can just upgrade to the Digium package without moving or changing any config? There is nothing specific about the packages that is going to make this situation any better or worse than any method of upgrading from Asterisk 1.6.X to Asterisk 1.8. Issues related to version compatibility can be found in the UPGRADE*.txt files in the Asterisk source. http://svn.asterisk.org/view/asterisk/trunk/UPGRADE-1.8.txt?view=markup -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.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
Re: [asterisk-users] Checking status of a cell phone
On Mon, 28 Mar 2011 14:12:09 +0200, magnu...@inputinterior.se wrote: Its not the Avaya that makes the call back, it is mobile. I thought the way you handled things, is that Asterisk would call your cellphone through the Avaya PBX just to check whether the cellphone is in_use/busy. At what point does the cellphone call Avaya or Asterisk back? -- _ -- 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] DTMF input while waiting in queue...
Wow... completely missed that. It was right there in the text. Sorry and thanks Sherwood! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: Monday, March 28, 2011 11:07 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF input while waiting in queue... On 3/28/2011 7:54 AM, Louis Carreiro wrote: Hey all! I'm trying to figure out how to have a queue accept an inbound caller's key press to action on. At first I'm just trying to implement a Press 1 to leave a voice mail announced and at any time in the queue, the user can press 1 and go to the queue's voicemail. Later I'd like to have it accept Press 1 if this is an x issue, press 2 if this a y problem and I'll have UserEvent's generated for the press. *snip* In your queues.conf, in the definition for 1820, add the following: context=queue1820-exit Then, in your dialplan create a new context: [queue1820-exit] exten = 1,1,Noop(Caller Pressed 1 to leave a voicemail) exten = 1,n,Voicemail(voicemailbox,theoptionsyouwant) exten = 1,n,Hangup That should get you started...Read about the context configuration option here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf Cheers! -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions 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 -- _ -- 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] Checking status of a cell phone
I was a little unclear, it is not the cell phone that does the call-back, it is the cell-phone-network. We can define 3 traffic-cases per cell-phone: 1) If cell-phone wont anser in x seconds call number a. 2) if cell-phone is busy call number b. 3) if cell-phone is unavailable call number c. From ami, a set db entry 0733025975 = 0 (Idle) from ami, make a short call (1 second) to 0733025975 wait 0.5 second check the db entry for 0733025975 when i wait for 0.5 second and my cell phone is busy, i will get a call to number b I catch that call in dialplan and set 0733025975 = 1 (InUse) Ofc, if cell-phone is unavailable, i will get call to number c I catch that call in dialplan and set 0733025975 = 4 (Unavailable) -Ursprungligt meddelande- From: Gilles Sent: Monday, March 28, 2011 10:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Checking status of a cell phone On Mon, 28 Mar 2011 14:12:09 +0200, magnu...@inputinterior.se wrote: Its not the Avaya that makes the call back, it is mobile. I thought the way you handled things, is that Asterisk would call your cellphone through the Avaya PBX just to check whether the cellphone is in_use/busy. At what point does the cellphone call Avaya or Asterisk back? -- _ -- 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
