[asterisk-users] call count per peer
Hello, Is there easiest way to count active channels per peer (peername) using AMI or CLI? The hard way would be : check all active channel and get their information : sip show channel 24609-fk-016ebea9-6e1e59...@10.10.10.1 and we get Peername . Thanks -- Best Regards, Giedrius -- _ -- 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] Astribank problem
On Mon, Feb 01, 2010 at 07:42:51AM +, frangky robert wrote: I do some test: 1.unplug usb connector from server to astricon 2.unplug power to astricon 3.plug-in the power to astricon 4.plug-in the usb connector Here is the log from /var/log/messages after doing the 1st step. Feb 1 19:38:24 localhost last message repeated 2 times Feb 1 19:43:39 localhost kernel: ERR-xpp_usb: xusb-0 (usb-:00:1d.7-3) [X1038295]: nonzero write bulk status received: -71 (pending_writes=1) Feb 1 19:43:39 localhost kernel: usb 2-3: USB disconnect, address 3 Feb 1 19:43:39 localhost kernel: ERR-xpp_usb: XBUS-00: xusb_listen: usb_submit_urb failed: -19 Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Disconnecting Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Deactivating Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Release XPDS Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-00: Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-10: Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-20: Remove Feb 1 19:43:39 localhost kernel: NOTICE-xpp: worker_reset: worker(XBUS-00)-xpds_init_done=0 Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Atribank Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Astribank Release Feb 1 19:43:39 localhost kernel: INFO-xpp_usb: xusb-0 (usb-:00:1d.7-3) [X1038295]: now disconnected Feb 1 19:43:39 localhost 'astribank_hook'[3728]: offline(XBUS-00): 0/1 from /etc/dahdi/xpp_order Feb 1 19:43:39 localhost 'astribank_hook'[3735]: All Astribanks offline And, this is the log after doing 4th step. Feb 1 19:44:20 localhost kernel: usb 2-3: new high speed USB device using ehci_hcd and address 4 Feb 1 19:44:20 localhost kernel: usb 2-3: configuration #1 chosen from 1 choice Feb 1 19:44:21 localhost 'xpp_fxloader'[3847]: Exiting... XPP_HOTPLUG_DISABLED Seems like you explicitly disabled firmware loading by setting XPP_HOTPLUG_DISABLED in /etc/dahdi/init.conf . Just rem-out that line. lsusb result is: [r...@localhost ~]# lsusb Bus 002 Device 004: ID e4e4:1160 Bus 002 Device 001: ID : Bus 006 Device 001: ID : Bus 006 Device 002: ID 04b3:3025 IBM Corp. Bus 004 Device 001: ID : Bus 008 Device 001: ID : Bus 007 Device 001: ID : Bus 001 Device 001: ID : Bus 005 Device 001: ID : Bus 003 Device 001: ID : here is the msg when i do /usr/share/dahdi/xpp_fxloader [r...@localhost ~]# /usr/share/dahdi/xpp_fxloader usb This only runs USB firmware loading. And as the firmware loading is explicitly disabled on your system, the FPGA firmware will still not get loaded. This is also something that you would have seen if you would run 'dahdi_hardware -v' So basically just remove that line from init.conf and replug the Astribank. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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
Re: [asterisk-users] connect problem unless when verbose
On 02/01/2010 08:38 AM, randall wrote: hi all, just had a terrible and sleepless weekend at the office trying to get asterisk going, its just tough love ;) have tried several asterisk versions but i currently have the following setup on debian lenny that kind of works. asterisk-1.6.2.0 dahdi-linux-complete-2.2.0.2+2.2.0 libpri-1.4.10.2 freepbx-2.6.0 setting up the sip devices is no problem at all, the difficulty i have is setting up 6xisdn2 lines with 2xb410p cards. besides the fact that i have no clue about what i'm doing i find the available documentation very very confusing, but i finally managed to make outgoing calls to my mobile this morning, sort off. when calling my mobile i hear a ringtone on my sip device and my mobile actually rings, YEAH!!! however, when i accept the call on my mobile my sip device keeps on ringing and my mobile gives no sound at all, when cancelling the call it simply cancels. except, and this i don't understand, i issue asterisk -rv (only with the v option), then i can connect and talk to myself, i often talk to myself when i spent a weekend at the office but this time its justifiable ;) anybody has a clue what could trigger this behavior??? , update!!! apparently it sometimes does work, randomly, guess the -v was a very lucky shot, i repeated it 20 times. i do seem to get this message everytime a connection fails [Feb 1 09:25:14] ERROR[2867] chan_dahdi.c: XXX Message longer than it should be?? XXX after applying this patch below the problem seem to have dissapeared for now https://issues.asterisk.org/view.php?id=16048 -- _ -- 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 for productive Calling Card System
Dear List, i have been thinking of building a calling cards solution based on Asterisk and a2billing.. i have a few questions regarding this solution and was hoping you may have the answers and could be generous enough to offer them. the servers i'm thinking of are with the following Specs: Processor: Intel X3210 Ram: 8Gb HDD: 2x500 GB Sata Internet Link: 100mbps Dedicated was thinking of using one for Database and the other for SIP trunking and calling card purposes. my questions are: 1- from your experience .. would a server with the previous specs handle a pressure of 200 or more outbound calls and 200 inbound (from access numbers)? what are the approximate concurrent call count supported by this hardware? 2- do you suggest using 64bit Centos or other OS? 3- for such usage what codecs do you prefere? 711u? 723? gsm? knowing that we are after good quality calls. my experiences are with small call centers up to 40 seats .. Thank you for your help and support. -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. http://clk.atdmt.com/GBL/go/201469226/direct/01/ -- _ -- 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] sip to dahdi and billsec
It entirely depends on the technology used to interface to the PSTN. You have not specified what technology/hardware you are using to connect to the PSTN. For instance if you are using POTS(plain old telephone service - analog copper fed lines), you do not get answer supervision back from the telco.-- I am using tdm400 card with one fxo port. I am using analog line so I guess it's POTS analog cooper fed line. So it is impossible to distinguish ringing from talking and billsec must start when ringing begin due missing answer supervision from telco ? Thanks for reply Use Free Software http://www.fsf.org/ --- Four essential software freedoms: 1) To study source code 2) To copy program 3) To modify source code 4) To redistribute modified program under condition that new user has all 4 freedoms. Richard M. Stallman -- _ -- 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] connect problem unless when verbose
On Mon, Feb 01, 2010 at 08:38:36AM +0100, randall wrote: hi all, just had a terrible and sleepless weekend at the office trying to get asterisk going, its just tough love ;) have tried several asterisk versions but i currently have the following setup on debian lenny that kind of works. asterisk-1.6.2.0 dahdi-linux-complete-2.2.0.2+2.2.0 libpri-1.4.10.2 freepbx-2.6.0 setting up the sip devices is no problem at all, the difficulty i have is setting up 6xisdn2 lines with 2xb410p cards. besides the fact that i have no clue about what i'm doing i find the available documentation very very confusing, but i finally managed to make outgoing calls to my mobile this morning, sort off. when calling my mobile i hear a ringtone on my sip device and my mobile actually rings, YEAH!!! however, when i accept the call on my mobile my sip device keeps on ringing and my mobile gives no sound at all, when cancelling the call it simply cancels. You try to connect two devices, ISDN and SIP. Both have their own complexities. I would sugest that you start by breaking this into two: first make sure an incoming ISDN call can make it into your PBX. e.g. into a simple IVR. Also make sure you can call your phone from Asterisk: In the Asterisk CLI: originate SIP/your-peer-name Application Playback demo-instruct Or: originate SIP/your-peer-name Application Echo -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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
Re: [asterisk-users] forward call back up same trunk to external cell phone problem
Hi- can anyone help with this. I'm really stuck as apparently it should work. Is it a problem with the ITSP, with using the same trunk for both legs of the call etc? John On 30 January 2010 08:57, John Taylor j...@vetsurgeon.org.uk wrote: Hi If I have an incoming call coming down a SIP trunk to a particular internal SIP extension- I can answer the extension fine, all works well However, if I change extension.conf from dialling the internal extension to forward the call to an external cell phone (up the same trunk as the incoming leg of the call) I cannot get any audio and get the following error message on the console: [Jan 30 08:38:42] WARNING[27575]: rtp.c:1145 ast_rtp_read: RTP Read too short i.e. change from [voipfone_incoming] exten = s,1,Dial(SIP/203,20,t) to [voipfone_incoming] exten = s,1,Dial(SIP/07123123...@voipfone,20,t) What's wrong?! John -- _ -- 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 1.2.37 + BLF + ParkedCalls + SPA962
On 26 January 2010 04:21, Joel Lansden j...@digitalparadise.net wrote: Greetings all. First off, thank you for your time on this. I have spent literally 12 hours searching every forum and article I can find, and I’m going cross-eyed, so I need to bother everyone with this. I am running * 1.2.37, and I am trying to get the hints working, so I can turn one of my SPA962’s LED’s red when someone parks a call. I have used Button #3 on my SPA962 to successfully monitor Zap channels, SIP channels, trunks.. basically, everything ELSE, so I am pretty sure my phone is set up properly. But I CANNOT get this puppy to show me when someone parks a call, regardless of the fact that call parking also works perfectly on this system. I have found so much conflicting information regarding the [parkedcalls] context, and the hints entry for extensions.conf (to use “exten=701,hint,park:7...@parkedcalls” vs. “exten=701,hint,Local/7...@parkedcalls”), as well as how to set up features.conf, I just don’t know which end is up. Does anyone out there know how to make this work with my setup? When I type “show hints”, earlier is was saying “State:Unavailable”, now it shows “State:Idle” but never does anything when I park a call. Here’s the files: [snip] AFAIK, asterisk 1.2.37 does not contain the required code to BLF monitor parked calls. IIRC You need the metermaid patch. Hope that helps, Steve -- _ -- 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] Odd error mssage on DAHDI lines
hi, you can lookup the causes in the sources check you dahdi-configuration (especially the groups...) is there everything ok? what does dahdi_tools or the other cli-commands say, that give you information about the available channels? yves /* Causes for disconnection (from Q.931) */ #define AST_CAUSE_UNALLOCATED1 #define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2 #define AST_CAUSE_NO_ROUTE_DESTINATION 3 #define AST_CAUSE_CHANNEL_UNACCEPTABLE 6 #define AST_CAUSE_CALL_AWARDED_DELIVERED7 #define AST_CAUSE_NORMAL_CLEARING16 #define AST_CAUSE_USER_BUSY17 #define AST_CAUSE_NO_USER_RESPONSE18 #define AST_CAUSE_NO_ANSWER19 #define AST_CAUSE_CALL_REJECTED21 #define AST_CAUSE_NUMBER_CHANGED22 #define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27 #define AST_CAUSE_INVALID_NUMBER_FORMAT 28 #define AST_CAUSE_FACILITY_REJECTED29 #define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY30 #define AST_CAUSE_NORMAL_UNSPECIFIED31 #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34 #define AST_CAUSE_NETWORK_OUT_OF_ORDER 38 #define AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41 #define AST_CAUSE_SWITCH_CONGESTION42 #define AST_CAUSE_ACCESS_INFO_DISCARDED 43 #define AST_CAUSE_REQUESTED_CHAN_UNAVAIL44 #define AST_CAUSE_PRE_EMPTED45 #define AST_CAUSE_FACILITY_NOT_SUBSCRIBED 50 #define AST_CAUSE_OUTGOING_CALL_BARRED 52 #define AST_CAUSE_INCOMING_CALL_BARRED 54 #define AST_CAUSE_BEARERCAPABILITY_NOTAUTH 57 #define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL 58 #define AST_CAUSE_BEARERCAPABILITY_NOTIMPL 65 #define AST_CAUSE_CHAN_NOT_IMPLEMENTED 66 #define AST_CAUSE_FACILITY_NOT_IMPLEMENTED 69 #define AST_CAUSE_INVALID_CALL_REFERENCE81 #define AST_CAUSE_INCOMPATIBLE_DESTINATION 88 #define AST_CAUSE_INVALID_MSG_UNSPECIFIED 95 #define AST_CAUSE_MANDATORY_IE_MISSING 96 #define AST_CAUSE_MESSAGE_TYPE_NONEXIST 97 #define AST_CAUSE_WRONG_MESSAGE 98 #define AST_CAUSE_IE_NONEXIST99 #define AST_CAUSE_INVALID_IE_CONTENTS 100 #define AST_CAUSE_WRONG_CALL_STATE 101 #define AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE 102 #define AST_CAUSE_MANDATORY_IE_LENGTH_ERROR 103 #define AST_CAUSE_PROTOCOL_ERROR111 #define AST_CAUSE_INTERWORKING127 Richard Kenner schrieb: What's this: -- Attempting call on DAHDI/g1/9removed for application Wait(5) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- Channel 0/2, span 1 got hangup, cause 44 -- Forcing restart of channel 0/2 on span 1 since channel reported in use -- Hungup 'DAHDI/2-1' Where can I look up cause 44. And if this is the sort of transient error that seems to be implied by the Forcing restart message, why isn't it retried? -- _ -- 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] Stuck logger rotation
Any idea what can cause this? asterisk*CLI core show channels Channel Location State Application(Data) Logger/rotates...@default:1 Down(None) 1 active channel 0 active calls 20229 calls processed asterisk*CLI core show chan channel channels channeltypes channeltype asterisk*CLI core show channel Logger/rotate -- General -- Name: Logger/rotate Type: NULL UniqueID: 1265014922.40051 Caller ID: (N/A) Caller ID Name: (N/A) DNID Digits: (N/A) Language: en State: Down (0) Rings: 0 NativeFormats: 0x0 (nothing) WriteFormat: 0x0 (nothing) ReadFormat: 0x0 (nothing) WriteTranscode: No ReadTranscode: No 1st File Descriptor: -1 Frames in: 0 Frames out: 0 Time to Hangup: 0 Elapsed Time: 3h47m49s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (None) Blocking in: (Not Blocking) Variables: filename=/var/log/asterisk/messages CDR Variables: level 1: dst=s level 1: dcontext=default level 1: channel=Logger/rotate level 1: start=2010-02-01 04:02:02 level 1: duration=13668 level 1: billsec=0 level 1: disposition=NO ANSWER level 1: amaflags=DOCUMENTATION level 1: uniqueid=1265014922.40051 I have a process: 7630 ?S 0:00 gzip -9 /var/log/asterisk/messages.2 The directory looks wierd: [r...@asterisk asterisk]# ls -l messages* -rw-rw-rw- 1 root root 0 Feb 1 04:02 messages.0 -rw-rw-rw- 1 root root 0 Feb 1 04:02 messages.1 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.1.1 -rw-rw-rw- 1 root root 0 Feb 1 04:02 messages.1.1.1 -rw-rw-rw- 1 root root 0 Feb 1 04:02 messages.1.1.1.1 -rw-rw-rw- 1 root root 3101914 Jan 31 23:14 messages.1.1.1.1.1 -rw-rw-rw- 1 root root 5407775 Jan 1 00:08 messages.1.1.1.2 -rw-rw-rw- 1 root root 0 Feb 1 04:02 messages.1.1.2 -rw-rw-rw- 1 root root 0 Jan 1 04:02 messages.1.1.2.1 -rw-rw-rw- 1 root root 914542 Dec 1 01:38 messages.1.1.3 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.1.2 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.1.2.1 -rw-r--r-- 1 root root 0 Jan 1 04:02 messages.1.2.1.1 -rw-r--r-- 1 root root 0 Dec 1 04:02 messages.1.2.2 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.1.3 -rw-r--r-- 1 root root 0 Jan 1 04:02 messages.1.3.1 -rw-r--r-- 1 root root 774669 Nov 1 01:10 messages.1.4 -rw-rw-rw- 1 root root 0 Feb 1 04:02 messages.2 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.2.1 -rw-rw-rw- 1 root root 0 Jan 1 04:02 messages.2.1.1 -rw-r--r-- 1 root root 0 Dec 1 04:02 messages.2.2 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.2.gz -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.2.gz.1 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.2.gz.1.1 -rw-r--r-- 1 root root 0 Jan 1 04:02 messages.2.gz.1.1.1 -rw-r--r-- 1 root root 0 Dec 1 04:02 messages.2.gz.1.2 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.2.gz.2 -rw-r--r-- 1 root root 0 Jan 1 04:02 messages.2.gz.2.1 -rw-r--r-- 1 root root 31 Nov 1 04:02 messages.2.gz.3 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.3 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.3.1 -rw-r--r-- 1 root root 0 Jan 1 04:02 messages.3.1.1 -rw-r--r-- 1 root root 0 Dec 1 04:02 messages.3.2 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.4 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.4.1 -rw-r--r-- 1 root root 0 Jan 1 04:02 messages.4.1.1 -rw-r--r-- 1 root root 0 Nov 1 04:02 messages.4.2 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.5 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.5.1 -rw-r--r-- 1 root root 0 Jan 1 04:02 messages.5.1.1 -rw-r--r-- 1 root root 0 Dec 1 04:02 messages.5.2 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.6 -rw-r--r-- 1 root root 0 Jan 1 04:02 messages.6.1 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.7 -rw-r--r-- 1 root root 0 Jan 1 04:02 messages.7.1 -rw-r--r-- 1 root root 0 Feb 1 04:02 messages.8 -rw-r--r-- 1 root root 169055 Oct 1 00:10 messages.9 -- _ -- 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] NVFaxDetect
Hi all, Do anyone has a detailed procedure for NV_application install? I have search as I was told, but I did no find any thing accurate. Thanks ML -- _ -- 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] mysterious rippled sound with IAX
Hi all, I know this could sound like a ghost story but... I have prepared an Asterisk 1.4.26.2 PBX on a Debian Lenny, the same I always prepare, same hw, same sw. I connected it to 2 iax phones using a hub: it works! I take everything it to our customer place, same PBX, same phones, same hub, same cables: the audio is rippled! Changing phones, cables and hub gives the same result. It does not happen with SIP phones but unfortunately I cannot use SIP protocol. :( Tried with a trixbox 2.6.2.2 (Asterisk 1.4.22) on a normal laptop and it works. It's like there something in our customer environment which is ruining IAX sound, maybe interfering with the PBX hardware (elettromagnetic waves?!?). I know it seems illogical or not scientific but this is what happens. Has anybody ever encountered a bizarre problem like this? Thank you. Giorgio -- _ -- 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] Set CDR userfield for Queues
1.4.x doesn't natively support logging the queue to to mysql, and can't find
any way without batching it from the /var/log/asterisk/queue
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
Sent: Saturday, January 30, 2010 8:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set CDR userfield for Queues
Thereis an option with log queue stored into database. You can do an
cross join from cdr and queue log using callid. With this solution you
can track your call over your asterisk system.
I wrote about this in old post and submit an complete solution.
Regards,
On Sun, Jan 24, 2010 at 1:14 PM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
Yeah, after hours of trying Friday, I got working by a macro.. I didn't
like
the outcome using a context, the macro cdr records looked cleaner.
[macro-queue]
exten = s,1,Answer()
exten = s,n,Queue(${ARG1}|rn)
exten = s,n,Set(MEMBERINTERFACE='NOANSWER')
exten = s,n,VoiceMail(${ARG1},u)
exten = s,n,Hangup()
exten = h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
Which is called from:
exten = _?,1,Macro(queue,${EXTEN})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deep D
Sent: Friday, January 22, 2010 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set CDR userfield for Queues
I just added a line with 'h'extension.
My dialplan is like this
[mycontext]
exten = s,1,Queue(6000)
exten = h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
On Sat, Jan 23, 2010 at 12:14 AM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
setinterfacevar=yes
Needs to be under each queue
What does your dialplan end up looking like?
I would like to add to mine, and stop running a cron job..
exten = 5000,1,Answer
exten = 5000,n,Queue(5000|rn)
exten = 5000,n,VoiceMail(5000,u)
exten = 5000,n,Hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deep D
Sent: Friday, January 22, 2010 1:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set CDR userfield for Queues
The 'h' extension worked. Thanks.
The other option of 'memebermacro' did not work. On the asterisk
console I could see that the macro is executed and cdr userfield is
set when agent answers the call, but the userfield doesn't show up in
the generated cdr.
Also I had one more question. Doesn't setinterfacevar=yes work when
it is declared in the general section? I had to declare it for each
queues.
On Fri, Jan 22, 2010 at 10:37 PM, Carlos Chavez cur...@telecomabmex.com
wrote:
On Fri, 2010-01-22 at 20:25 +0530, Deep D wrote:
I want to do something like this
exten = 1234,n,Queue(6000,c)
exten = 1234,n,Set(CDR(userfield)=${Agent}) ;; where Agent is the
agent who answered the call
exten = 1234,n,Hangup
Actually because the user will hangup within the Queue
application
you
cannot do that. You will have to use the h extension to make the change
to the userfield. Something like this:
h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
Make sure you have setinterfacevar=yes in your queue.conf so that
variable is created when the user is connected to the agent. Another
possibility is to run a macro by using membermacro=somemacro and set
the userfield within that macro. I think that option is only available
on Asterisk 1.6.X and not for older ones though. You can also run an
AGI script (you can set it as an option in the Queue commando) that will
set the userfield as this AGI is run just before the call is bridged to
the agent but the ${MEMBERINTERFACE} is already set.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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[asterisk-users] Problems with recordings of call using Monitor
Hi,
I'm using the default Asterisk function Monitor to record calls, but i have
some issue's with this, the problem is when a call is finished, it never mix
in out together, bellow you can see my call configuration:
exten = _8.,1,Monitor(wav,${EXTEN},m)
exten = _8.,n,Dial(SIP/${EXTEN:1...@${exten:1})
(the 8 prefix is due to testing of the system)
The reason you see the ex...@exten is because of OpenSips, it's connected to
Asterisk, and some of my users i would like to record are behind opensips
and reachable by dialing ext@domain but in sip.conf i defined the host,
that's why i'm using ex...@exten.
Even on a normal Asterisk machine, i have issue's with recording, i'm using
Asterisk 1.6.2.
Anybody got any tips on this?
Thanks,
Peter
--
Groet // Kind regards,
Peter den Hartog
Sent from Amsterdam, NH, Netherlands
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Re: [asterisk-users] Set CDR userfield for Queues
Yes, over asterisk 1.4.x you need put an cron or deamon to load queue
log into database. I suggest check on the list for more details.
Regards,
On Mon, Feb 1, 2010 at 8:51 AM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
1.4.x doesn't natively support logging the queue to to mysql, and can't find
any way without batching it from the /var/log/asterisk/queue
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
Sent: Saturday, January 30, 2010 8:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set CDR userfield for Queues
Thereis an option with log queue stored into database. You can do an
cross join from cdr and queue log using callid. With this solution you
can track your call over your asterisk system.
I wrote about this in old post and submit an complete solution.
Regards,
On Sun, Jan 24, 2010 at 1:14 PM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
Yeah, after hours of trying Friday, I got working by a macro.. I didn't
like
the outcome using a context, the macro cdr records looked cleaner.
[macro-queue]
exten = s,1,Answer()
exten = s,n,Queue(${ARG1}|rn)
exten = s,n,Set(MEMBERINTERFACE='NOANSWER')
exten = s,n,VoiceMail(${ARG1},u)
exten = s,n,Hangup()
exten = h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
Which is called from:
exten = _?,1,Macro(queue,${EXTEN})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deep D
Sent: Friday, January 22, 2010 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set CDR userfield for Queues
I just added a line with 'h'extension.
My dialplan is like this
[mycontext]
exten = s,1,Queue(6000)
exten = h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
On Sat, Jan 23, 2010 at 12:14 AM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
setinterfacevar=yes
Needs to be under each queue
What does your dialplan end up looking like?
I would like to add to mine, and stop running a cron job..
exten = 5000,1,Answer
exten = 5000,n,Queue(5000|rn)
exten = 5000,n,VoiceMail(5000,u)
exten = 5000,n,Hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deep D
Sent: Friday, January 22, 2010 1:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set CDR userfield for Queues
The 'h' extension worked. Thanks.
The other option of 'memebermacro' did not work. On the asterisk
console I could see that the macro is executed and cdr userfield is
set when agent answers the call, but the userfield doesn't show up in
the generated cdr.
Also I had one more question. Doesn't setinterfacevar=yes work when
it is declared in the general section? I had to declare it for each
queues.
On Fri, Jan 22, 2010 at 10:37 PM, Carlos Chavez cur...@telecomabmex.com
wrote:
On Fri, 2010-01-22 at 20:25 +0530, Deep D wrote:
I want to do something like this
exten = 1234,n,Queue(6000,c)
exten = 1234,n,Set(CDR(userfield)=${Agent}) ;; where Agent is the
agent who answered the call
exten = 1234,n,Hangup
Actually because the user will hangup within the Queue
application
you
cannot do that. You will have to use the h extension to make the change
to the userfield. Something like this:
h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
Make sure you have setinterfacevar=yes in your queue.conf so that
variable is created when the user is connected to the agent. Another
possibility is to run a macro by using membermacro=somemacro and set
the userfield within that macro. I think that option is only available
on Asterisk 1.6.X and not for older ones though. You can also run an
AGI script (you can set it as an option in the Queue commando) that will
set the userfield as this AGI is run just before the call is bridged to
the agent but the ${MEMBERINTERFACE} is already set.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] MATH
There's nothing wrong with this per se; it just needs to be in a context;
try it this way;
- exten = 8284,1,Goto(domath,s,1)
[domath]
Exten = s,1,play(to-call-num-press)
- exten = 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
- exten = 4,n,WaitExten(3)
- exten = 4,n,Goto(domath,s,1)
- exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
- exten = 2,n,Waitexten(3)
- exten = 2,n,Goto(domath,s,1)
- exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
- exten = 3,n,WaitExten(3)
- exten = 3,n,Goto(domath,s,1)
- exten = 9,1,SayNumber(${TOTAL})
- exten = 9,n,Play(vm-goodbye)
- exten = 9,n,hangup
--
Danny Nicholas
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Sunday, January 31, 2010 5:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MATH
On 31/01/10 6:27 PM, Thomas Perron wrote:
what is wrong with this please:
;exten = 4,1,WaitExten(3)
exten = 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 4,n,WaitExten(3)
exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten = 2,n,Waitexten(3)
exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten = 3,n,WaitExten(3)
exten = 9,1,SayNumber(${TOTAL})
Heh, you might need to say what you're expecting and what you're getting :D
Straight off, all I can see is that 2 does 200, 3 does 300 and 4 does 500.
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Cheers,
Matt Riddell
Managing Director
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Re: [asterisk-users] sip to dahdi and billsec
Uros Djokic wrote: It entirely depends on the technology used to interface to the PSTN. You have not specified what technology/hardware you are using to connect to the PSTN. For instance if you are using POTS(plain old telephone service - analog copper fed lines), you do not get answer supervision back from the telco.-- I am using tdm400 card with one fxo port. I am using analog line so I guess it's POTS analog cooper fed line. So it is impossible to distinguish ringing from talking and billsec must start when ringing begin due missing answer supervision from telco ? Thanks for reply Answer supervision is missing in one sense. It was never part of the spec for this type of telco line. You will have to forgive calls under x number of seconds in duration as if they never occured or get a different type of connection from your telco that will include answer supervision. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] MATH
This is my feeble attempt at explaining this;
DTMF is processed by two dial-plan commands that I'm familiar with (there
are more but I'm trying to speak from my knowledge)
Waitexten reads 1 or more DTMF digits and goes to an extension in the
dialplan.
Example
- exten = s,1,playback(welcome)
- exten = s,n,Waitexten(10,m)
- exten = 0,1,verbose(operator)
- exten = 100,1,verbose(call extension 100)
- exten = 200,1,verbose(call extension 200)
The other command is read - example
- exten = 101,1,read(foobar,message,5,skip,1,9)
- exten = 101,n,verbose(you entered ${foobar})
- exten = 101,n,playback(vm-goodbye)
- exten = 101,n,hangup
HTH
--
Danny Nicholas
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch
Sent: Sunday, January 31, 2010 2:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MATH
Clue,
If a caller keys in 4 5 3 will some variable return 453?
I ASSume yes, since you can make menu selections with DTMF, obviously you
can process the results further or in other ways than that.
Cary
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Sunday, January 31, 2010 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MATH
On Sun, 31 Jan 2010, Thomas Perron wrote:
does dtmf any any variable that i can capture and use w/ some logic
like in the case of a gotoif
Anyone have a clue what this means? Anyone? Anyone?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] MATH
Thank you.
I was also thinking of using the READ application to store dtmp variabes.
Then total them up at the end.
More to follow.
P
On Mon, Feb 1, 2010 at 9:20 AM, Danny Nicholas da...@debsinc.com wrote:
There's nothing wrong with this per se; it just needs to be in a context;
try it this way;
- exten = 8284,1,Goto(domath,s,1)
[domath]
Exten = s,1,play(to-call-num-press)
- exten = 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
- exten = 4,n,WaitExten(3)
- exten = 4,n,Goto(domath,s,1)
- exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
- exten = 2,n,Waitexten(3)
- exten = 2,n,Goto(domath,s,1)
- exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
- exten = 3,n,WaitExten(3)
- exten = 3,n,Goto(domath,s,1)
- exten = 9,1,SayNumber(${TOTAL})
- exten = 9,n,Play(vm-goodbye)
- exten = 9,n,hangup
--
Danny Nicholas
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Sunday, January 31, 2010 5:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MATH
On 31/01/10 6:27 PM, Thomas Perron wrote:
what is wrong with this please:
;exten = 4,1,WaitExten(3)
exten = 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 4,n,WaitExten(3)
exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten = 2,n,Waitexten(3)
exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten = 3,n,WaitExten(3)
exten = 9,1,SayNumber(${TOTAL})
Heh, you might need to say what you're expecting and what you're getting :D
Straight off, all I can see is that 2 does 200, 3 does 300 and 4 does 500.
--
Cheers,
Matt Riddell
Managing Director
___
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Re: [asterisk-users] MATH
Here's how I would do this based on the post below (it's below in outlook
express)
Exten = 866,1,Goto(tommath,s,1)
[tommath]
- exten = s,1,Read(NUMBER,instruct,2,skip,5)
- exten = s,n,Gotoif($[${NUMBER} = 1]?one)
- exten = s,n,Gotoif($[${NUMBER} = 2]?two)
- exten = s,n,Gotoif($[${NUMBER} = 3]?three)
- exten = s,n,Gotoif($[${NUMBER} = 20]?done)
- exten = s,playback(system) - error message
- exten = s,n,Goto(tommath,s,1)
- exten = s,n(one),Set(TOTAL=${MATH(${TOTAL}+500,int)})
- exten = s,n,Goto(tommath,s,1)
- exten = s,n(two),Set(TOTAL=${MATH(${TOTAL}+200,int)})
- exten = s,n,Goto(tommath,s,1)
- exten = s,n(three),Set(TOTAL=${MATH(${TOTAL}+300,int)})
- exten = s,n,Goto(tommath,s,1)
- exten = s,n(done),SayNumber(${TOTAL})
- exten = s,n,playback(vm-goodbye)
- exten = s,n,hangup
Regards,
Danny Nicholas
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
Sent: Saturday, January 30, 2010 7:48 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MATH
I want to create a script for IVR that compiles responses, aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the results
= 500+200+300 =1000
then,
exten = s,n,Read(NUMBER,,1000)
exten = s,n,SayDigits(${NUMBER})
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Re: [asterisk-users] NVFaxDetect
On 02/01/10 09:54, Mariano Lecuona wrote: Hi all, Do anyone has a detailed procedure for NV_application install? I have search as I was told, but I did no find any thing accurate. Thanks ML I was running it with Asterisk-1.2 (nice application) one would think that it should be incorporate into asterisk but they are not doing it, nor the developer is interesting in upgrading the code ever-time there is a new version of asterisk and something is broken. I was not able to install it with version 1.4 -- Joseph -- _ -- 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] NVFaxDetect
Joseph wrote: I was running it with Asterisk-1.2 (nice application) one would think that it should be incorporate into asterisk but they are not doing it, nor the developer is interesting in upgrading the code ever-time there is a new version of asterisk and something is broken. I was not able to install it with version 1.4 It has never been offered for inclusion into Asterisk, so they cannot do it. The Asterisk project does not 'pull in' code, it must be specifically offered for inclusion by the author(s)/copyright holder(s) of the code. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.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
Re: [asterisk-users] NVFaxDetect
On 02/01/10 12:01, Kevin P. Fleming wrote: Joseph wrote: I was running it with Asterisk-1.2 (nice application) one would think that it should be incorporate into asterisk but they are not doing it, nor the developer is interesting in upgrading the code ever-time there is a new version of asterisk and something is broken. I was not able to install it with version 1.4 It has never been offered for inclusion into Asterisk, so they cannot do it. The Asterisk project does not 'pull in' code, it must be specifically offered for inclusion by the author(s)/copyright holder(s) of the code. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org Thank for explanation, it clears things up. It is/was a nice application and I'm just wandering why develp a code and let it it die -- Joseph -- _ -- 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] NVFaxDetect
YMMV, but NVFaxdetect can be activated in 1.4.x; Try this link http://blogtech.oc9.com/index.php?option=com_contentview=articleid=77%3A20 071121astItemid=8 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Monday, February 01, 2010 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] NVFaxDetect On 02/01/10 12:01, Kevin P. Fleming wrote: Joseph wrote: I was running it with Asterisk-1.2 (nice application) one would think that it should be incorporate into asterisk but they are not doing it, nor the developer is interesting in upgrading the code ever-time there is a new version of asterisk and something is broken. I was not able to install it with version 1.4 It has never been offered for inclusion into Asterisk, so they cannot do it. The Asterisk project does not 'pull in' code, it must be specifically offered for inclusion by the author(s)/copyright holder(s) of the code. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org Thank for explanation, it clears things up. It is/was a nice application and I'm just wandering why develp a code and let it it die -- Joseph -- _ -- 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 -- _ -- 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 with recordings of call using Monitor
On Mon, Feb 1, 2010 at 8:55 AM, Peter den Hartog peterdenhar...@gmail.com wrote: I'm using the default Asterisk function Monitor to record calls, but i have some issue's with this, the problem is when a call is finished, it never mix in out together, bellow you can see my call configuration: Perhaps you would prefer to use MixMonitor() rather than Monitor() -- _ -- 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] disable comfort noise
This is how I understand it. The other end is trying to set up comfort noise and asterisk is letting you know that it's trying to do so and maybe you can turn this off on the other end. I have a particular voip provider where I get this message. I think if you get it turned off there's a little bit better performance on the connection. By now people are getting used to calls without comfort noise but for a long time it threw people off because they weren't sure if the call was still connected. On 1/30/2010 1:30 AM, uzzi wrote: On Fri, Jan 29, 2010 at 1:14 PM, ad...@3a.hu mailto:ad...@3a.hu wrote: To get back to the original poster's possible situation, i've seen this with my first IP phone, which was a cisco 7912 (SIP image). With that phone, asterisk sometimes gave me this same error. I'm quite sure i've asked the very same question here back then (probably i was a bit more specific :). Since it is related to only this type of phone, i've gone to different ip phone products. regards adam -- _ -- 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 Please correct me if I'm wrong As the error says, Please turn off on client if possible. Comfort noise (aka silent suppression, or Voice Activity Detection (VAD)) is not supported by Asterisk. It needs to be turned off on the user (client) end. This may be a phone or another switch/PBX. See http://www.voip-info.org/wiki/view/RTP+Silence+Suppression for more details -- _ -- 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] One way audio with Grandstream HT503
Hello list ! I'm having one way audio on incoming and outgoing calls. Outgoing audio works fine, incoming audio is not working. My setup is the following : incoming calls : PSTN -- FXOport -- HT503 -- WANport -- Asterisk -- WANport -- HT503 (the same) -- FXSport -- DECTphone outgoing calls : DECTphone -- FXSport -- HT503 -- WAN-port -- Asterisk -- internet (VoIPprovider) I've done a TCPdump on the Asterisk-server : 18:20:21.189504 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.193065 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.210111 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.213065 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.229848 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.233064 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.250013 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.253049 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.269737 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.273058 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.289918 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.293048 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.310080 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.313058 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.329819 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.333047 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.349985 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.353054 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.370164 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.373046 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.389886 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.393031 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.410053 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.413042 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.430218 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.433033 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.449957 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.453039 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.469694 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.473039 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.489857 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.493059 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.509593 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.513028 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.530190 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.533039 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.549930 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.553025 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.573037 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.593024 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.613011 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.630162 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.630208 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.633022 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.653012 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.670492 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.673019 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.689799 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.689844 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.693028 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.713036 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.730131 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.730176 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.733147
Re: [asterisk-users] beroNet BN4S0e PCI Express ISDN Card with chan_dahdi
On 31/01/2010 04:35, Tzafrir Cohen wrote: Yes, please see https://issues.asterisk.org/view.php?id=16493 Basically the driver needs minimal fixing. Probably just to add the PCI ID to the list. Hi, Thanks Gonna have a look at it. -- _ -- 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] (no subject)
Please descard me from the asterisk users list...thanks (Abu Nasar Mahmud) -- _ -- 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] Use a BLF for monitoring
Is there a way to make a virtual extension busy programmatically? I want to be able to turn lights on and off on a Polycom phone from a script. -Matt -- _ -- 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] Use a BLF for monitoring
Is there a way to make a virtual extension busy programmatically? I want to be able to turn lights on and off on a Polycom phone from a script. That's what the Custom device type is for. -- _ -- 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] Use a BLF for monitoring
Richard Kenner wrote: Is there a way to make a virtual extension busy programmatically? I want to be able to turn lights on and off on a Polycom phone from a script. That's what the Custom device type is for. please elaborate I would like to know too -- _ -- 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] Use a BLF for monitoring
In your dialplan, you should put in...sth like exten = 1001,hint,Custom:virtext1001 In your script, you should put in...sth like Set(DEVSTATE(Custom:virtext001=INUSE); Set(DEVSTATE(Custom:virtext001=NOTINUSE); In the phone directory.xml, define an entry with ct=1001 and turn bw on. Reboot phone of course. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of jon pounder Sent: Tuesday, 2 February 2010 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Use a BLF for monitoring Richard Kenner wrote: Is there a way to make a virtual extension busy programmatically? I want to be able to turn lights on and off on a Polycom phone from a script. That's what the Custom device type is for. please elaborate I would like to know too -- _ -- 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 -- _ -- 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] (no subject)
If you read your message all the way to the end, and every posting, you will discover exactly how to do that on your own. asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users nasar mahmud wrote: Please descard me from the asterisk users list...thanks (Abu Nasar Mahmud) Checked by AVG - www.avg.com Version: 9.0.733 / Virus Database: 271.1.1/2660 - Release Date: 01/31/10 14:35:00 -- Dog is my co-pilot -- _ -- 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] Use a BLF for monitoring
That's what the Custom device type is for. please elaborate I would like to know too See http://www.voip-info.org/wiki/view/Asterisk+func+device_State -- _ -- 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] Smallest possible Asterisk VM
How small can an Asterisk system be, in terms of disk space utilized? I am looking for just asterisk, with mysql, postgresql, or sqlite, with PHP and Python. After finishing the build and removing the tools, how small can the whole system be? 100Mb, 200Mb? Can packages be used to build the whole system, like using debs and rpms alone? /vfclists -- _ -- 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] Smallest possible Asterisk VM
I think Astlinux comes in under 100MB. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com Twitter: http://www.twitter.com/bschorr Facebook: http://www.facebook.com/rolandschorr -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Frank Church Sent: Monday, February 01, 2010 19:41 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Smallest possible Asterisk VM How small can an Asterisk system be, in terms of disk space utilized? I am looking for just asterisk, with mysql, postgresql, or sqlite, with PHP and Python. After finishing the build and removing the tools, how small can the whole system be? 100Mb, 200Mb? Can packages be used to build the whole system, like using debs and rpms alone? /vfclists -- __ ___ -- 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 -- _ -- 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] Smallest possible Asterisk VM
On Tue, Feb 2, 2010 at 6:41 AM, Frank Church voi...@googlemail.com wrote: How small can an Asterisk system be, in terms of disk space utilized? I am looking for just asterisk, with mysql, postgresql, or sqlite, with PHP and Python. AskoziaPBX is 30MB installed. It has Asterisk 1.6.1, nearly all of the modules and PHP. No external db support yet, and no need for python so far. It only has the minimum prompt set to cover what functionality is in the GUI: - http://www.askozia.com The biggest portion of your space is going to go to media. Music-on-hold can be limited to a single file but to have a complete set of prompts, it will cost you another 10MB installed or so. I think a truly minimal Asterisk install with externally stored media would come in at less than 20MB installed. If you optimize the kernel even further for your specific hardware and take out all unneeded user space utils, it should be less than 15MB installed. Hope that helps, -Michael -- _ -- 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] Smallest possible Asterisk VM
AstLinux is well under that. You could build a custom image that contains only what you want and have it under 30M. We have support for sqlite3, but not mysql or postgresql. You would have to build your own package to include python. Our build environment is based on buildroot, but has been heavily modified to suit our needs. Feel free to ask questions on the astlinux-devel mailing list. http://www.astlinux.org Darrick On 02/02/2010 12:02 AM, Ben Schorr wrote: I think Astlinux comes in under 100MB. Ben M. Schorr -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Frank Church Sent: Monday, February 01, 2010 19:41 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Smallest possible Asterisk VM How small can an Asterisk system be, in terms of disk space utilized? I am looking for just asterisk, with mysql, postgresql, or sqlite, with PHP and Python. After finishing the build and removing the tools, how small can the whole system be? 100Mb, 200Mb? Can packages be used to build the whole system, like using debs and rpms alone? /vfclists -- _ -- 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] Realtime queue strategy issue
hi,all all realtime queue work fine except one thing: in the queue_table ,when i change strategy from ringall to linear need asterisk to restart! [Feb 2 15:41:51] WARNING[4106]: app_queue.c:1532 queue_set_param: Changing to the linear strategy currently requires asterisk to be restarted. i use asterisk 1.6.2.1, is this a bug of Asterisk 1.6.2.1? does anyone have some idea for me ? Thanks! -- Best regards, Sucan -- _ -- 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
