[asterisk-users] wctdm24xx IRQ missing
Hello all, I just instaled a tdm2400 Digium card on my asterisk box. When it boots, I can see some error messages in dmesg. wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 8 ms in order to compensate. wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 10 ms in order to compensate. wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 11 ms in order to compensate. wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 12 ms in order to compensate. wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 13 ms in order to compensate. I try to change the IRQ of the card, using the setpci command, without success. I am using dadhi 2.2.1 and asterisk 1.4.24 Has anyone the same problem? How could I change to fix this errors? Any clue will be wellcomed. Voipcrazy. -- _ -- 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 Log viewer
Hello, I want to analyze the asterisk logs files, looking for all kind of errors, ¿Anyboby knows any asterisk logs analyzer? Thanks all, Voipcrazy -- _ -- 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] Switchboad like application
Hello all, Anybody could point me any clue about an Open Source or licensed switchboard for my users? ARI or FOP is not enought for my users. Thanks in advance. VoipCrazy -- _ -- 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] RTP ports
Hello, I need to limit the RTP ports used by an asterisk in a client, Actualy the range defined is from 1 to 2 udp ports. If I only have 10 local sip extension ¿how many ports/range should I set up in /etc/asterisk/rtp.conf? Which is the way to calculate the rtp ports needed on an instalation? Thanks in advance, Voipcrazy. -- _ -- 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] Snom Provisioning
Hello all, I've to deploy about 200 snom320 phones on a instalation. Do you know any knid of tool to help me with this amount of phones? I'm thinking in a provisioning tool which I use for setting up the phones. Any clue would be welcomed. Thanks. Voip-Crazy -- _ -- 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] Billing applications
Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. True CDR, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Billing applications
Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. - True CDR. Better that asterisk one, With suport for transfers - I do not need support for reseller - Billing for Voip, PSTN trunks I need a light app. I'm not searching a heavy app. with a lots of modules and applicacions. I need a ligth application for a soho and its needs. Any one are using a billing application which fits this needs? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] onnecting two asterisk using B410p BRI cards
I just plug the junper in NT mode with no success. VoipCrazy 2009/8/15 Paul Hales pdha...@optusnet.com.au: Use a standard network cable - but you have to activate the 'terminate' jumper on the NT end. - Also, the new BRI stuff in dahdi is much easier to work with than misdn. PaulH voip crazy wrote: Hello all, I'm trying to conect two asterisk servers using two B410p Digium cards. One card on each server. I just setting up the first BRI port on server A as nt_ptp and the first BRI port on server B as te_ptp. I use an ethernet wire to connect the first port of server A (nt_ptp) with the first port on server B (te_ptp) but the port light cotinues blinking on red on both sides once the cable was pluged. Then I use an isdn crossover wire with this king of schema and the lights get blinking red again. Tx+ 3 --+ +- 3 . X Rx+ 4 --+ +- 4 . Tx- 5 --+ +--5 . X Rx- 6 --+ +--6 In both servers when I do in asterisk CLI misdn shos stacks, the port one on each machine shows Server A: BEGIN STACK_LIST: * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Server B: BEGIN STACK_LIST: * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Which kind of cable should I use? Why both in ports L1Link is failed? How could I solve that? Any clue will be welcomed. Thanks in advance. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] onnecting two asterisk using B410p BRI cards
Hello all, I'm trying to conect two asterisk servers using two B410p Digium cards. One card on each server. I just setting up the first BRI port on server A as nt_ptp and the first BRI port on server B as te_ptp. I use an ethernet wire to connect the first port of server A (nt_ptp) with the first port on server B (te_ptp) but the port light cotinues blinking on red on both sides once the cable was pluged. Then I use an isdn crossover wire with this king of schema and the lights get blinking red again. Tx+ 3 --+ +- 3 .X Rx+ 4 --+ +- 4 . Tx- 5 --+ +--5 .X Rx- 6 --+ +--6 In both servers when I do in asterisk CLI misdn shos stacks, the port one on each machine shows Server A: BEGIN STACK_LIST: * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Server B: BEGIN STACK_LIST: * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Which kind of cable should I use? Why both in ports L1Link is failed? How could I solve that? Any clue will be welcomed. Thanks in advance. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Printing faxes
Hello list, I have an asterisk / hylafax / iaxmodem configured in one machine. All is working nicely. Now I need the fax to be print when arriving. ¿Anybody have this feature implementing in their systems? ¿How is the best way to get that? Any clue will be welcomed. Thanks. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Webcall app needed
Hello all, I need to configure an application which let me to call from a web page. Someone has experience using apps to make webcalls? Which software do you use? Thanks. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hylafax asterisk iaxmodem problem
Hello all, I have an asterisk box running in a customer with Hylafax, iaxmodem, asterisk 1.2.18. The service can receive faxes, from a lot of fax machines, but there are a couple of them that asterisk Hylafax cannot complete. This calls arrive the asterisk box, asterisk detect that this calls are fax, asterisk answer the call, and then Hangup the call. But hylafax do not receive nothing. When I run zap show channel 1, on the asterisk CLI. The outpuit shows, File Descriptor: 20 Span: 2 Extension: Dialing: no Context: from-zaptel Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook Why some faxes do not get received? What could be wrong? Any clue wil be welcomed. Thanks in advanced. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WebCall application
Hello list, Does anybody know any free WebCall solution to let our customer call us directly via our web site? Any clue will be welcomed. Thanks. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] B410p question
Hello list, I have got an asterisk box installed working ok with an b410p card to make and receive isdn calls. All works ok, but when a call is answer and the person starts to speak, always I can ear a beep during the call. This beep is ear some times in about 30 seconds between each beep. Pasted bellow I send /etc/misdn-init.conf and /etc/asterisk/misdn.conf Any clue will be apreciated. Thanks. VoipCrazy - My /etc/misdn-init.conf - # # Configuration file for your misdn hardware # # Usage: /usr/sbin/misdn-init start|stop|restart|config|scan|help # # # Card Settings # # Syntax: card=number,type[,option...] # #number count your cards beginning with 1 #type either 0x1,0x4 or 0x8 for your hfcmulti hardware, # or the name of your card driver module. #option ulaw - uLaw (instead of aLaw) # dtmf - enable DTMF detection on all B-channels # # pcm_slave - set PCM bus into slave mode #If you have a set of cards, all wired via PCM. Set #all cards into pcm_slave mode and leave one out. #The left card will automatically be Master. # # ignore_pcm_frameclock - this can be set in conjunction with # pcm_slave. If this card has a # PCI Bus Position before the Position # of the Master, then this port cannot # yet receive a frameclock, so it must # ignore the pcm frameclock. # # rxclock- use clocking for pcm from ST Port # crystalclock - use clocking for pcm from PLL (genrated on board) # watchdog - This dual E1 Board has a Watchdog for #transparent mode # # card=1,0x4 # # Port settings # # Syntax: port_type=port_number[,port_number...] # #port_typete_ptp - TE-Mode, PTP # te_ptmp - TE-Mode, PTMP # te_capi_ptp - TE-Mode (capi), PTP # te_capi_ptmp- TE-Mode (capi), PTMP # nt_ptp - NT-Mode, PTP # nt_ptmp - NT-Mode, PTMP #port_number port that should be considered # #te_ptmp=1,2,3,4 #te_ptmp=1,2 te_ptp=1,2,3,4 # # Port Options # # Syntax: option=port_number,option[,option...] # #option master_clock - use master clock for this S/T interface # (only once per chip, only for HFC 8/4) # optical - optical (only HFC-E1) # los - report LOS (only HFC-E1) # ais - report AIS (only HFC-E1) # slip - report SLIP (only HFC-E1) # nocrc4- turn off crc4 mode use double frame instead # (only HFC-E1) # # The master_clock option is essential for retrieving and transmitting # faxes to avoid failures during transmission. It tells the driver to # synchronize the Card with the given Port which should be a TE Port and # connected to the PSTN in general. # option=1,master_clock #option=2,ais,nocrc4 #option=3,optical,los,ais,slip # # General Options for your hfcmulti hardware # # poll=number # #Only one poll value must be given for all cards. #Give the number of samples for each fifo process. #By default 128 is used. Decrease to reduce delay, increase to #reduce cpu load. If unsure, don't mess with it!!! #Valid is 32, 64, 128, 256. # # dsp_poll=number # This is the poll option which is used by mISDN_dsp, this might # differ from the one given by poll= for the hfc based cards, since # they can only use multiples of 32, the dsp_poll is dependant on # the kernel timer setting which can be found in the CPU section # in the kernel config. Defaults are there either 100Hz, 250Hz # or 1000Hz. If your setting is either 1000 or 250 it is compatible # with the poll option for the hfc chips, if you have 100 it is # different and you need here a multiple of 80. # The default is to have no dsp_poll option, then the dsp itself # finds out which option is the best to use by itself # # pcm=number # #Give the id of the PCM bus. All PCM busses with the same ID #are expected to be connected and have equal slots. #Only one chip of the PCM bus must be master, the others slave. # # debug=number # #Enable debugging (see hfc_multi.h for debug options). # # dsp_options=number # # set this to 2 and you'll have software bridging instead of # hardware bridging. # # # dtmfthreshold=milliseconds # # Here you can tune the sensitivity of the dtmf tone recognizer. # # timer=1|0 # # set this to 1 if you want
[asterisk-users] Asterisk Queue question
When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy Fewest Calls working for a couple of mouths, and a new agent has been added this week in the queue and he is receiving all the incomings calls. How could I solve that? Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip Header Help
Dear List: I need to make a sip phone (spa942) answer a call but the phone must no ring. The user only has to show the callerId on the phone screen without any sound. How could I make that in asterisk? I tried to use Sip headers but I do not know how must I say the phone don't ring when received, only shows the callerID of the call. How could I do that with sip header? Which sip header should I send the phone to change the callerID of the call? Do you know any other way to ghet that. Any clue will be wellcomed. Thanks for your answer. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Thank you Hatem, I will try it now Thanks VoipCrazy 2008/9/2 hatem moiz [EMAIL PROTECTED]: you can do the following in sip .conf file register = username:[EMAIL PROTECTED] and after that write the configuration for the user: [ user ] username = host = qualify = secret = and so on, do this in the first of sip.conf file Best Regards On Mon, Sep 1, 2008 at 11:32 AM, voip crazy [EMAIL PROTECTED] wrote: Hatem, I cannot understan exactly what you told me. Could you try to explain that in other words. Better if you could post an example of this SIP trunk. thanks in advance. Voip Crazy 2008/9/1 hatem moiz [EMAIL PROTECTED]: Asterisk is looking for a SIP trunk if you have recorded the usage of SIP trunks all it need is to find 1 SIP trunk, To fix your problem make a local sip trunk i mean sip trunk to 127.0.0.1 and make sure that it is the first one in sip.conf file. OR you can make a sip trunk to ATA in the same lan and also be sure that it is the first trunk in sip.conf . On Mon, Sep 1, 2008 at 9:58 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway errors
Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Igor, From asterisk, when internet is down I can ping all extensions. The same occurs in others instalations, when the internet is down, my lical extensions log off from asterisk. VoipCrazy 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Hatem, I cannot understan exactly what you told me. Could you try to explain that in other words. Better if you could post an example of this SIP trunk. thanks in advance. Voip Crazy 2008/9/1 hatem moiz [EMAIL PROTECTED]: Asterisk is looking for a SIP trunk if you have recorded the usage of SIP trunks all it need is to find 1 SIP trunk, To fix your problem make a local sip trunk i mean sip trunk to 127.0.0.1 and make sure that it is the first one in sip.conf file. OR you can make a sip trunk to ATA in the same lan and also be sure that it is the first trunk in sip.conf . On Mon, Sep 1, 2008 at 9:58 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outgoing calls
Hello list, How could I limit the outgoing calls for one trunks easily? Thanks VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco vs Asterisk
Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? Thanks. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk dimensioning
Hello all, I need to install asterisk for 900 sip users with 2 PRI ports. It is posible to handle this number of calls/extensions with only one asterisk machine? Which is the best way to install that? two asterisk with openser. One asterisk with openser . Is it necesary run a SER server on this enviroment? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dimensioning
Maybe 400 calls at one time. By the momento there aren`t voip trunks maybe in the future. About cluster, Which cluster solution will could be good option? Which solution could I use to do load balancing between two asterisk machines? Thanks again. Voipcrazy 2008/7/9 Tom Moore [EMAIL PROTECTED]: How many calls do you expect to be going at one time? Do you have any sip trunks for the users to call out on? Unless this ratio really works for you I'm not sure a 15 to 1 ratio works for most people. I wouldn't just depend on a single server for this purpose. I'll leave it to the cluster guys to describe the ideal setup you should use. I have an idea of how I might do it, but I wouldn't want to get it wrong. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of voip crazy Sent: Wednesday, July 09, 2008 3:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk dimensioning Hello all, I need to install asterisk for 900 sip users with 2 PRI ports. It is posible to handle this number of calls/extensions with only one asterisk machine? Which is the best way to install that? two asterisk with openser. One asterisk with openser . Is it necesary run a SER server on this enviroment? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Removing voicemail messages
Hello, I want to create an script which remove all the old voicemail messages. I make a simple Bash script to delete all the new messages for the extension 100. Something like, rm /var/spool/asterisk/voicemail/defaul/100/INBOX Should I update any index file or something after reemove them? Thanks in advance VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager proxy
Hello all, Some one is using asterisk and queuemetrics connected via astmanproxy? How about your experience? Which proxy do you use in this kind of connection? In my instalation asterisk and Queuemetrics are installed on diferent machines and I want to avoid manager problems Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softphone accepting sip messages
Hello all, Someone knows any softphone which accept messages using sipsak? I just tried X-Lite and portsip without success Thanks Voipcrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfers with TE12xp
Hello all, I have an asterisk PBX working perfectly, and the transfers between extensions, works ok. The problem, when I receive a call from the line connected to the TE12Xp, and I try to transfer it, the calls hangs up. I have other analog lines and I can tranfer all the without problems. I've pasted the zapata config for the PRI line, please tell me what could be wrong and the cause my calls hangs up. Any clue will be welcomend. Best Regards. VoipCrazy -- /etc/asterisk/zapata.conf --- language=es context=from-zaptel relaxdtmf=yes signalling=pri_cpe signallingtype=euroisnd rxwink=300 ; Atlas seems to use long (250ms) winks ;usedistinctiveringdetection=yes callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 ;callgroup=1 ;pickupgroup=1 immediate=no ;busydect=yes busycount=6 faxdetect=both group=0 channel=1-15,17-31 - ___ -- 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] Transfers with TE12xp
More info about the problem. This occurs, when I try to transfer using the *2 funcionality into aterisk Thanks 2008/6/16 voip crazy [EMAIL PROTECTED]: Hello all, I have an asterisk PBX working perfectly, and the transfers between extensions, works ok. The problem, when I receive a call from the line connected to the TE12Xp, and I try to transfer it, the calls hangs up. I have other analog lines and I can tranfer all the without problems. I've pasted the zapata config for the PRI line, please tell me what could be wrong and the cause my calls hangs up. Any clue will be welcomend. Best Regards. VoipCrazy -- /etc/asterisk/zapata.conf --- language=es context=from-zaptel relaxdtmf=yes signalling=pri_cpe signallingtype=euroisnd rxwink=300 ; Atlas seems to use long (250ms) winks ;usedistinctiveringdetection=yes callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 ;callgroup=1 ;pickupgroup=1 immediate=no ;busydect=yes busycount=6 faxdetect=both group=0 channel=1-15,17-31 - ___ -- 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] Dial command and its g option
I need to execute an action after a call is hangup. I just see the command Dial has an option for that, the g option. I configure the dial command as exten = s,n,Dial(SIP/100,100,Ttg) How should I add the line which the command will be executed after the dial command in this example? I don`t how its works, someone could put a example about the way to use it. Thanks you in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI after Hangup
Which is the way to run an AGI after hangup a call? The problem I have is when the call dies the AGI dies too I try the Dial command g option, but it does not work for me Any clue will be welcomed. Thanks VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI after Hangup
Thanks for your answers, DeadAGI was the solution. Thanks again. Voipcrazy 2008/6/12 Andrea Cristofanini [EMAIL PROTECTED]: You have to run DeadAGI, in h . Regards Andrea Cristofanini voip crazy ha scritto: Which is the way to run an AGI after hangup a call? The problem I have is when the call dies the AGI dies too I try the Dial command g option, but it does not work for me Any clue will be welcomed. Thanks VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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
[asterisk-users] PRI error cause hangup calls
Dear all, When I make a call using my PRI line, all goes well, but suddently the call hangs up. I searched the asterisk logs, and I found that. Write to 55 failed: Unknown error 500 Short write: 0/15 (Unknown error 500) What does this mean? Why this occurs? How could I solve that? Someone could tell me if it was a primary error (the primary shows red alert in all its channels) or it could be a driver or config problem? Thanks in advance. VoipCrazy. Mar 27 14:28:00 DEBUG[20313] chan_zap.c: Got event Alarm(4) on channel 1 (index 0) Mar 27 14:28:00 VERBOSE[20313] logger.c: Write to 55 failed: Unknown error 500 Mar 27 14:28:00 VERBOSE[20313] logger.c: Short write: 0/15 (Unknown error 500) Mar 27 14:28:00 WARNING[20313] chan_zap.c: Detected alarm on channel 1: Red Alarm Mar 27 14:28:00 DEBUG[20313] chan_zap.c: disabled echo cancellation on channel 1 Mar 27 14:28:00 DEBUG[20313] channel.c: Didn't get a frame from channel: Zap/1-1 Mar 27 14:28:00 DEBUG[20313] channel.c: Bridge stops bridging channels SIP/7008-b6a158e0 and Zap/1-1 Mar 27 14:28:00 DEBUG[20313] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Mar 27 14:28:00 DEBUG[20313] chan_zap.c: Hangup: channel: 1 index = 0, normal = 36, callwait = -1, thirdcall = -1 Mar 27 14:28:00 DEBUG[20313] chan_zap.c: disabled echo cancellation on channel 1 Mar 27 14:28:00 DEBUG[20313] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Mar 27 14:28:00 DEBUG[20313] chan_zap.c: Updated conferencing on 1, with 0 conference users Mar 27 14:28:00 DEBUG[20313] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 Mar 27 14:28:00 DEBUG[20313] chan_zap.c: disabled echo cancellation on channel 1 Mar 27 14:28:00 VERBOSE[20313] logger.c: -- Hungup 'Zap/1-1' Mar 27 14:28:00 DEBUG[20313] app_dial.c: Exiting with DIALSTATUS=ANSWER. Mar 27 14:28:00 VERBOSE[20313] logger.c: == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/7008-b6a158e0' in macro 'dialout-trunk' Mar 27 14:28:00 VERBOSE[20313] logger.c: == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/7008-b6a158e0' Mar 27 14:28:00 VERBOSE[20313] logger.c: -- Executing Macro(SIP/7008-b6a158e0, hangupcall) in new stack Mar 27 14:28:00 VERBOSE[20313] logger.c: -- Executing ResetCDR(SIP/7008-b6a158e0, w) in new stack Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 2: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 2 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 3: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 3 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 4: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 4 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 5: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 5 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 6: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 6 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 7: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 7 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 8: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 8 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 9: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 9 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 10: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 10 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 11: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 11 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 12: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 12 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 13: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 13 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 14: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 14 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 15: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 15 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 17: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 17 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 18: Red Alarm Mar 27
[asterisk-users] Sending SMS
Hello all, I want to send SMS using asterisk, I just read there are lot of apps to do that, but I do not know which to choose, like cmd SMS, Fast SMS, ZIM-SMS,.etc. http://www.voip-info.org/wiki/view/SMS Which is the way you use to send SMS messages using asterisk? Which apps do you use? Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird Zaptel sound after anwser calls
Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a Clack. I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes callerid=asreceived transfer=yes callreturn=yes rxgain=-3.0 txgain=-3.0 immediate=no busydetect=yes busycount=8 answeronpolarityswitch=yes hanguponpolarityswitch=yes ringtimeout=8000 faxdetect=both group=0 signalling=fxs_ks channel = 7 think the problem is not by echo cause I use fxotune, and the problem persist. I made lots of TDM02B instalations and never get this kind of problem. Any clue will be welcomed. Thanks in advance. Regards VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird Zaptel sound after anwser calls
I forgot to say that I'm using bristuff-0.4.0 with zaptel 1.4.4, libpri 1.4.1 and asterisk 1.4.9 Thanks. 2008/2/22, voip crazy [EMAIL PROTECTED]: Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a Clack. I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes callerid=asreceived transfer=yes callreturn=yes rxgain=-3.0 txgain=-3.0 immediate=no busydetect=yes busycount=8 answeronpolarityswitch=yes hanguponpolarityswitch=yes ringtimeout=8000 faxdetect=both group=0 signalling=fxs_ks channel = 7 think the problem is not by echo cause I use fxotune, and the problem persist. I made lots of TDM02B instalations and never get this kind of problem. Any clue will be welcomed. Thanks in advance. Regards VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware needed
Dear List, I have to plan an instalation of an asterisk box for over 400 extensions (Sip and Iax2) and 4 PRI channels. I do not know which hardware (server) should I buy to support this amount of extensions. Someone made a similar instalation? which hardware (server) did you use? Which was the processor type and the amount of memory used by the server? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attendant phone
Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let me do that? Which is the maximun number of extensions your phones can monitor and which models phones are? Thanks, VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and fax
Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that Hylafax/Iaxmodem. And the Asterfax solution does dislike cause its licensing. The TE420B, is configured in E1 mode. Which is the best solution to use with this hardware? Which solution do you use to send an receibe fax? Thanks VoIPCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and fax
I want to receibe the fax via mail and send faxes via web interface and a digital send and receibe fax list. Voipcrazy 2008/2/13, Giorgio Incantalupo [EMAIL PROTECTED]: Hi VoIPCrazy, why don't you use an ATA device such as Grandstream 486 or similar? Giorgio Incantalupo voip crazy wrote: Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that Hylafax/Iaxmodem. And the Asterfax solution does dislike cause its licensing. The TE420B, is configured in E1 mode. Which is the best solution to use with this hardware? Which solution do you use to send an receibe fax? Thanks VoIPCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- 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
[asterisk-users] PRI with 20 channels
Dear all, I have got a PRI line with E1 20 channels, my question is: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 If this is my zaptel config for an E1 PRI line, Which would be the config for a reduced PRI line for 15 channels? and for 20 channels? Thanks in advance. VoIPcrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Qsig link
Hello all, I need to conect an Asterisk with an Alcatel OmniPBX 4400 using an E1 port. It is the first time I make this kind of connection and I do not know exactly how to get it working. Someone has experience with this kind of connection? Could you paste a zapata.con and zaptel.conf files with QSIG configuration? Any clue will be wellcomed. Thanks Voipcrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Qsig link
Thank you Rob for your soon answer. I will tell you how it works Voipcrazy 2008/1/21, Rob Hillis [EMAIL PROTECTED]: Pretty easy actually. - zaptel.conf span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf -- usecallerid=yes hidecallerid=no echocancel=yes echocancelwhenbridged=yes echotraining=800 relaxdtmf=yes rxgain=-3.0 txgain=-6.0 busydetect=yes busycount=8 immediate=no switchtype=qsig group=0 signalling=pri_net channel = 1-15,17-31 You may need to set signalling to pri_cpe depending on how your Alcatel is configured. This is a working configuration from an Asterisk -- NEC NEAX7400 ICS system. (You may not need busy detection - the only reason I have it enabled is because there are a couple of analogue lines connected to the NEC without hangup detection) voip crazy wrote: Hello all, I need to conect an Asterisk with an Alcatel OmniPBX 4400 using an E1 port. It is the first time I make this kind of connection and I do not know exactly how to get it working. Someone has experience with this kind of connection? Could you paste a zapata.con and zaptel.conf files with QSIG configuration? Any clue will be wellcomed. Thanks Voipcrazy -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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
[asterisk-users] Snom 370 buton Recordings
Hello all, I am using the Snom 370 phone with firmware Snom370-SIP 7.0.17* *connected to an asterisk 1.2.14 and I can't record any calls using the Recording button on this phone. The extension I configured on this phone has the values Recording on demand, an the voicemail enabled. I am using FreePBX to manage my PBX. How should I configure the Function keys to make this work? Anybody have made this button works on this phone? How? Any clue will be welcomed. Thanks in advance. Voipcrazy * * ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Change the Voice promps in asterisk 1.4
Hello all, Which is the best way to change the default Voice promps in asteriosk 1.4from english to french? And if I would like to add a new Voice promp set, how is the way to do? Thanks in advance. VoipCrazy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom 320 with TDM02B and echo problems
Hello all, I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones, on an amd_64 processor. All goes well, the voice is clear on the remote side but in the Voip side, where the Snom 320 is placed, I hear my voice, but don't in the line, the echo is on the phone. I just play with zapata gain values and with the Snom mic volume, but the echos does not disapperars. the phone is updated to firmware 6.5.12, the last i have found. Any clue about how to eliminate de echo in the snom 320 phone? What could I do to solve that? Thanks in advance. VoipCrazy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to delete voice mail messages?
Hello all, Could I create a script to delete the first messages on my voice mail? In this script should I update any messages index file or there isn't any file to index them? Could you share any script to do that? Thanks in advance. VoipCrazy. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk hylafax iaxmodem
Witch interface are you using to send faxes, SIP, IAX, ZAP, MISDN,...,etc VoipCrazy 2007/10/17, Giedrius Augys [EMAIL PROTECTED]: CountryCode:1 AreaCode: 800 FAXNumber: +3705203230 LongDistancePrefix: 1 InternationalPrefix:011 DialStringRules:etc/dialrules ServerTracing: 0xFFF SessionTracing: 0xFFF RecvFileMode: 0600 LogFileMode:0600 DeviceMode: 0600 RingsBeforeAnswer: 1 SpeakerVolume: off GettyArgs: -h %l dx_%s LocalIdentifier:Giedrius Augys TagLineFont:etc/lutRS18.pcf TagLineFormat: Nuo %%l|%c|Page %%P of %%T MaxRecvPages: 200 # # # Modem-related stuff: should reflect modem command interface # and hardware connection/cabling (e.g. flow control). # ModemType: Class1 # use this to supply a hint # # Enabling this will use the hfaxd-protocol to set Caller*ID # #ModemSetOriginCmd: AT+VSID=%s,%d # # If glare during initialization becomes a problem then take # the modem off-hook during initialization, and then place it # back on-hook when done. # #ModemResetCmds:ATH1\nAT+VCID=1 # enables CallID display #ModemReadyCmds:ATH0 Class1AdaptRecvCmd: AT+FAR=1 Class1TMConnectDelay: 400 # counteract quick CONNECT response # # If you have trouble with V.17 receiving or sending, # you may want to enable one of these, respectively. # #Class1RMQueryCmd: !24,48,72,96 # enable this to disable V.17receiving #Class1TMQueryCmd: !24,48,72,96 # enable this to disable V.17sending # # You'll likely want Caller*ID display (also displays DID) enabled. # ModemResetCmds: AT+VCID=1 # enables CallID display # # The pty does not support changing parity. # PagerTTYParity: none # # If you are missing Caller*ID data on some calls (but not all) # and if you do not have adequate glare protection you may want to # not answer based on RINGs, but rather enable the CallIDAnswerLength # for NDID, disable AT+VCID=1 and do this: # #RingsBeforeAnswer: 0 #ModemRingResponse: AT+VRID=1 # Uncomment DATE and TIME if you really want them, but you probably don't. #CallIDPattern: DATE= #CallIDPattern: TIME= CallIDPattern: NMBR= CallIDPattern: NAME= CallIDPattern: ANID= #CallIDPattern: USER=# username provided by call #CallIDPattern: PASS=# password provided by call #CallIDPattern: CDID=# DID context in call CallIDPattern: NDID= #CallIDAnswerLength:4 2007/10/17, Jonn R Taylor [EMAIL PROTECTED]: Giedrius Augys wrote: Hi, I have problems with asterisk and hylafax+ iaxmodem. I can successfully send faxes to Panasonic KX-FT932 fax, but with Xerox WorkCentre M20i I have problems: No carrier. This is hylafax log, maybe you can suggest me where to find ... Oct 17 07:38:48.22: [22428]: SESSION BEGIN 00041 180037052390906 Oct 17 07:38:48.22: [22428]: HylaFAX (tm) Version 4.4.2 Oct 17 07:38:48.22: [22428]: SEND FAX: JOB 27 DEST 37052390906 COMMID 00041 DEVICE '/dev/ttyIAX0' FROM 'root [EMAIL PROTECTED]' USER root Oct 17 07:38:48.22: [22428]: STATE CHANGE: RUNNING - SENDING Oct 17 07:38:48.22: [22428]: -- [12:AT+FCLASS=1\r] Oct 17 07:38:48.22: [22428]: -- [2:OK] Oct 17 07:38:48.22 : [22428]: MODEM set XON/XOFF/FLUSH: input ignored, output disabled Oct 17 07:38:48.22: [22428]: DIAL 37052390906 Oct 17 07:38: 48.22: [22428]: -- [16:ATDT37052390906\r] Oct 17 07:39:30.86: [22428]: -- [10:NO CARRIER] Oct 17 07:39:30.86: [22428]: SEND FAILED: JOB 27 DEST 37052390906 ERR [2] No carrier detected Oct 17 07:39: 30.86: [22428]: SEND FAILED: JOB 27 DEST 37052390906 ERR [333] No carrier detected; too many attempts to dial Oct 17 07:39:31.86: [22428]: -- [5:ATH0\r] Oct 17 07:39:31.86: [22428]: -- [2:OK] Oct 17 07:39:31.86: [22428]: MODEM set DTR OFF Oct 17 07:39:31.86: [22428]: MODEM set baud rate: 0 baud (flow control unchanged) Oct 17 07:39:31.86: [22428]: STATE CHANGE: SENDING - MODEMWAIT (timeout 5) Oct 17 07:39:31.86: [22428]: SESSION END -- Pagarbiai / Best Regards, Giedrius Augys ___ --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 post your iaxmodem config file. Jonn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Eicon Diva and incoming Fax
Dear Armin,
This solve my problem, when I set softdtmf and relaxdtmf to off, my asterisk
machine starts to detect the incoming fax calls.
Thank for your help.
VoipCrazy.
2007/10/15, Armin Schindler [EMAIL PROTECTED]:
On Mon, 15 Oct 2007, voip crazy wrote:
Dear Armin,
Bellow I send you my /etc/asterisk/capi.conf file, I just set
faxdetect=both, but the card isn`t detect an incoming fax call.
I use capicommand(receivefax|...), and work well, but I need that
asterisk
or the diva card detects an incoming fax call to send it to a specific
context.
There are any way to use capicommand to detects fax incoming fax.
Don't set
softdtmf=on
relaxdtmf=on
because your DIVA card can do this with the onboard DSPs and fax detection
should work then too.
Armin
Thanks in advance.
VoipCrazy
--Capi.conf---
[general]
nationalprefix=0
internationalprefix=00
rxgain=1.0 ;linear receive gain (1.0 = no change)
txgain=1.0 ;linear transmit gain (1.0 = no change)
language=de ;set default language
;ulaw=yes;set this, if you live in u-law world instead of a-law
;jb. ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting
available.
;mohinterpret=default ;Asterisk 1.4: default music on hold class when
placed
on hold.
; interface sections ...
[ISDN] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;Use one interface section for each isdn port!
;ntmode=yes ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward
dial)
;when using NT-mode, 'DID' should be set in any case
incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * =
any
;defaultcid=123 ;set a default caller id to that interface for
dial-out,
;this caller id will be used when dial option 'd' is
set.
;controller=0;ISDN4BSD default
;controller=7;ISDN4BSD USB default
controller=1 ;capi controller number of this interface/port
group=1 ;dialout group
;prefix=0;set a prefix to calling number on incoming calls
softdtmf=on ;enable/disable software dtmf detection, recommended
for
AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
detection
faxdetect=both;enable faxdetection and redirection to EXTEN 'fax'
for
incoming and/or
;outgoing calls. (default='off', possible values:
'incoming','outgoing','both')
accountcode=Canal-RDSI ;PBX accountcode to use in CDRs
;amaflags=default;AMA flags for CDR ('default', 'omit', 'billing', or
'documentation')
context=from-pstn ;context for incoming calls
;holdtype=hold ;when the PBX puts the call on hold, ISDN HOLD will be
used. If
;set to 'local' (default value), no hold is done and the
PBX may
;play MOH.
holdtype=local
;immediate=yes ;DID: immediate start of pbx with extension 's' if no
digits were
; received on incoming call (no destination number
yet)
;MSN: start pbx on CONNECT_IND and don't wait for
SETUP/SENDING-COMPLETE.
; info like REDIRECTINGNUMBER may be lost, but this
is
necessary for
; drivers/pbx/telco which does not send SETUP or
SENDING-COMPLETE.
;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
;echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation (yes=g165)
;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
;echocancelold=yes;use facility selector 6 instead of correct 8
(necessary
for older eicon drivers)
;echotail=64 ;echo cancel tail setting (default=0 for maximum)
;echocancelnlp=1 ;activate non-linear-processing; this improves echo
cancel
ratio, but might
;incorporate variable gain in the signal path.
bridge=yes ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;PBX call group
;pickupgroup=1 ;PBX pickup group (which call groups are we allowed to
pickup)
;language=de ;set language for this device (overwrites default
language)
;disallow=all;RTP codec selection (valid with Eicon DIVA Server
only)
allow=all ;RTP codec selection (valid with Eicon DIVA Server only)
devices=2;number of concurrent calls (b-channels) on this
controller
;(2 makes sense for single BRI, 30/23 for PRI/T1)
;jb. ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting
available.
;mohinterpret=default ;Asterisk 1.4: default music on hold class when
placed
on hold.
;qsig=on ;enable use of Q.SIG extensions.
--EOF
[asterisk-users] Eicon Diva and incoming Fax
Hello all, I am trying to set up asterisk and hylafax to send and receibe fax. The machine is connected to the PSTN is an Eicon Diva BRI (1 BRI port). My problem is that , when I send a Fax from the PSTN to this machine, the asterisk or diva or hylafax, does not detect this call as a fax and asterisk answer that call like a voice call. How sould I configure the software modem (iaxmodem) to use with the Eicon Diva card? What sould I do to make the Eicon Diva detects an incoming fax? Thanks in advance. VoipCrazy. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Eicon Diva and incoming Fax
Dear Armin, the problem is my Eicon Diva Card does not detect aany fax-tone. Then the call is redirect as a voice call instead a fax call. How could I detect the fax.-tone with this kind of hardware? How could I enable receivefax? Thanks in advance. VoipCrazy 2007/10/15, Armin Schindler [EMAIL PROTECTED]: Hello VoipCrazy !? On Mon, 15 Oct 2007, voip crazy wrote: Hello all, I am trying to set up asterisk and hylafax to send and receibe fax. The machine is connected to the PSTN is an Eicon Diva BRI (1 BRI port). My problem is that , when I send a Fax from the PSTN to this machine, the asterisk or diva or hylafax, does not detect this call as a fax and asterisk answer that call like a voice call. How sould I configure the software modem (iaxmodem) to use with the Eicon Diva card? What sould I do to make the Eicon Diva detects an incoming fax? If asterisk shall not accept this call, then configure asterisk not to do so. Either use another number, or check the transfercapability (if the sender did set this correct). Why don't you receive the fax via asterisk? You can answer the call and if a fax-tone is detected, you switch to receivefax. Armin ___ --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] Eicon Diva and incoming Fax
Dear Armin,
Bellow I send you my /etc/asterisk/capi.conf file, I just set
faxdetect=both, but the card isn`t detect an incoming fax call.
I use capicommand(receivefax|...), and work well, but I need that asterisk
or the diva card detects an incoming fax call to send it to a specific
context.
There are any way to use capicommand to detects fax incoming fax.
Thanks in advance.
VoipCrazy
--Capi.conf---
[general]
nationalprefix=0
internationalprefix=00
rxgain=1.0 ;linear receive gain (1.0 = no change)
txgain=1.0 ;linear transmit gain (1.0 = no change)
language=de ;set default language
;ulaw=yes;set this, if you live in u-law world instead of a-law
;jb. ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting available.
;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed
on hold.
; interface sections ...
[ISDN] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;Use one interface section for each isdn port!
;ntmode=yes ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, 'DID' should be set in any case
incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
;defaultcid=123 ;set a default caller id to that interface for dial-out,
;this caller id will be used when dial option 'd' is set.
;controller=0;ISDN4BSD default
;controller=7;ISDN4BSD USB default
controller=1 ;capi controller number of this interface/port
group=1 ;dialout group
;prefix=0;set a prefix to calling number on incoming calls
softdtmf=on ;enable/disable software dtmf detection, recommended for
AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
detection
faxdetect=both;enable faxdetection and redirection to EXTEN 'fax' for
incoming and/or
;outgoing calls. (default='off', possible values:
'incoming','outgoing','both')
accountcode=Canal-RDSI ;PBX accountcode to use in CDRs
;amaflags=default;AMA flags for CDR ('default', 'omit', 'billing', or
'documentation')
context=from-pstn ;context for incoming calls
;holdtype=hold ;when the PBX puts the call on hold, ISDN HOLD will be
used. If
;set to 'local' (default value), no hold is done and the
PBX may
;play MOH.
holdtype=local
;immediate=yes ;DID: immediate start of pbx with extension 's' if no
digits were
; received on incoming call (no destination number yet)
;MSN: start pbx on CONNECT_IND and don't wait for
SETUP/SENDING-COMPLETE.
; info like REDIRECTINGNUMBER may be lost, but this is
necessary for
; drivers/pbx/telco which does not send SETUP or
SENDING-COMPLETE.
;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
;echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation (yes=g165)
;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary
for older eicon drivers)
;echotail=64 ;echo cancel tail setting (default=0 for maximum)
;echocancelnlp=1 ;activate non-linear-processing; this improves echo cancel
ratio, but might
;incorporate variable gain in the signal path.
bridge=yes ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;PBX call group
;pickupgroup=1 ;PBX pickup group (which call groups are we allowed to
pickup)
;language=de ;set language for this device (overwrites default language)
;disallow=all;RTP codec selection (valid with Eicon DIVA Server only)
allow=all ;RTP codec selection (valid with Eicon DIVA Server only)
devices=2;number of concurrent calls (b-channels) on this controller
;(2 makes sense for single BRI, 30/23 for PRI/T1)
;jb. ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting available.
;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed
on hold.
;qsig=on ;enable use of Q.SIG extensions.
--EOF--
2007/10/15, Armin Schindler [EMAIL PROTECTED]:
On Mon, 15 Oct 2007, voip crazy wrote:
Dear Armin,
the problem is my Eicon Diva Card does not detect aany fax-tone. Then
the
call is redirect as a voice call instead a fax call.
How could I detect the fax.-tone with this kind of hardware?
How could I enable receivefax?
Are we talking about a DIVA Server BRI card?
If yes, then the card can detect fax tone and you just need
to enabled this in capi.conf.
Then capicommand(receivefax|...) will help you.
Armin
2007/10/15, Armin Schindler [EMAIL PROTECTED]:
Hello
[asterisk-users] Virtual server Solution
Hello all, I'm looking for a solution to offer Virtual PBX, to my clients. I just saw software with multi-tenant support and I tested it, but no one likes me enought. Finally, I want to offer this service like a kind of hosting. Has you experience with multi-tenant software? Which has you tested? Has anyone experience about vhost, vserver, or something similar to run asterisk on it? Thanks VoIpCrazy ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Virtual server Solution
Dear Tzafrir, I just try Destar, but one thing I dislike was, that there are no posibilities to login the manager of each virtual PBX. Then customers cannot manage their owns PBX. VoiPCrazy 2007/9/24, Tzafrir Cohen [EMAIL PROTECTED]: On Mon, Sep 24, 2007 at 11:38:38AM +0200, voip crazy wrote: Hello all, I'm looking for a solution to offer Virtual PBX, to my clients. I just saw software with multi-tenant support and I tested it, but no one likes me enought. Finally, I want to offer this service like a kind of hosting. Has you experience with multi-tenant software? Which has you tested? Has anyone experience about vhost, vserver, or something similar to run asterisk on it? Try destar. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Hfcmulti and B410P Digium Card
Hello all, I am getting the following error in /var/log/syslog. I have got 2 B410P cards in this box. Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0153, z2=00d3) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0013, z2=0012) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0053, z2=0153) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0093, z2=0092) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=00d3, z2=0053) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0113, z2=0112) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0153, z2=00d3) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0013, z2=0012) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0053, z2=0153) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(2) reading 15 bytes (z1=004e, z2=0040) HDLC COMPLETE (f1=3, f2=2) got=15 Sep 19 17:13:31 localhost kernel: 02 01 06 06 08 01 63 5a 08 02 80 90 Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0093, z2=0092) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(2) has 383 bytes space left (z1=015d, z2=015d) sending 4 of 4 bytes HDLC I left untouched the /etc/init.d/misdn-init script to load the default values. Is needed the hfcmulti modules with this kind of cards? What is the menaing of this errors? Are something missconfigured? Any clue will be wellcomed Best Regards. VoipCrazy ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Hfcmulti and B410P Digium Card
Maybe I find the problem, It could be cause debug is enabled. Tomorrow I will change debug to disable and I will tell you the results. Regards. VoipCrazy 2007/9/19, voip crazy [EMAIL PROTECTED]: Hello all, I am getting the following error in /var/log/syslog. I have got 2 B410P cards in this box. Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0153, z2=00d3) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0013, z2=0012) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0053, z2=0153) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0093, z2=0092) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=00d3, z2=0053) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0113, z2=0112) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0153, z2=00d3) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0013, z2=0012) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0053, z2=0153) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(2) reading 15 bytes (z1=004e, z2=0040) HDLC COMPLETE (f1=3, f2=2) got=15 Sep 19 17:13:31 localhost kernel: 02 01 06 06 08 01 63 5a 08 02 80 90 Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0093, z2=0092) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(2) has 383 bytes space left (z1=015d, z2=015d) sending 4 of 4 bytes HDLC I left untouched the /etc/init.d/misdn-init script to load the default values. Is needed the hfcmulti modules with this kind of cards? What is the menaing of this errors? Are something missconfigured? Any clue will be wellcomed Best Regards. VoipCrazy ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Email to Voice
Hello all, Anyone knows any solution (Comercial or Free) to listen my email via a phone call? Thanks in advance. VoipCrazy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queues monitoring software
Hello all, A client of us, needs a queue monitoring system. In realtime he needs to now the PRI status, the agents logged in and logged out, the number of received calls by agent, ,etc. I am not a call center specialist and i want to find a call center software to offer to my client that fits his needs. I need a monitoring solution for incomming and outgoing calls and a queue management interface to create and/or modify queues or agents. Any one of you could has instalesd this kind of software? Which one? Which one could you recomend me? Thanks in advance. Voipcrazy. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hi ability solution
Hi all, On one of our client, I must to install an asterisk over a hi ability cluster. I have no experience with clusters an linux neither asterisk. Someone has installed an asterisk in a hi-ability enbviroment? How do you install the cluster? Witch solution did you use? Witch is the best cluster solution to use with asterisk? Thanks in advance, Voipcrazy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hi ability solution
I would say High Availability, sorry for my english. Any High availiability solution for asterisk? VoipCrazy 2007/6/25, Steve Totaro [EMAIL PROTECTED]: voip crazy wrote: Hi all, On one of our client, I must to install an asterisk over a hi ability cluster. I have no experience with clusters an linux neither asterisk. Someone has installed an asterisk in a hi-ability enbviroment? How do you install the cluster? Witch solution did you use? Witch is the best cluster solution to use with asterisk? Thanks in advance, Voipcrazy Do you mean High Ability, or High Availability I think Rocks is pretty good but I just started playing with it. I think it is more of a High Ability thing. http://www.rocksclusters.org/wordpress/ Thanks, Steve Totaro ___ --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
[asterisk-users] Astmanproxy
Hello all, Some of you are using astmanproxy with asttapi or activa TSP? How does you make to work? Thanks VoipCrazy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk TAPI interface
Hello, I need to connect asterisk 1.2.16, with a Contect Center software that works with TAPI. As I know, asterisk doesn't support TAPI directly, if needs a tirth party software. I just reading about asttapi and Activa TAPI. does anyone test this software? have you using asterisk againts a TAPI compatible software? Witch TAPI software do you test? Thanks in advance. VoipCrazy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Internet gateway problem
Hello all, I have got an asterisk server in my LAN, getting access to internet trought a router. I have observed in my asterisk box, when the internet connection in down, the phones can not register to my asterisk. It is like chan_sip, does not work without an internet connection. If when the router is down the telephones does not register, but when I type in my asterisk box route del default, teh phones then started to register against the asterisk. Why this is happenning? Why chan_sip, need a gateway or it does not start correctly? Why when I type route del default the phones started to register? Any clue will be wellcommed Thanks in advance VoipCrazy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nagios asterisk monitoring
Dear list,
I am trying to configure the nagios plugin called check_sip. I just read the
README file included with the plugin. I follow the readme steps to configure
the plugin, without success. In the nagios web interface I can see (No
output!) In the status information column. If I run the chech_sip plugin
from a linux console, I get
/usr/local/nagios/libexec# ./check_sip -u sip:[EMAIL PROTECTED]
SIP 200 OK: 0.00 second response time
I do not know why If I run the plugin from the consle it works ok, but if I
run it from Nagios web interface it does not run.
Anyone are using this plugin?
Could you helpme to solve that?
Any clue will be appreciated.
Thanks for your time.
VoipCrazy
Here goes my nagios check_sip plugin configuration.
define command{
command_namecheck_sip
command_line$USER1$/check_sip -u $ARG1$ -H $HOSTADDRESS$ -w 5
}
define service{
use generic-service
host_name -PBX
service_description SIP test
check_command check_sip!sip:[EMAIL PROTECTED]
contact_groups admins
max_check_attempts 4
normal_check_interval 5
retry_check_interval1
notification_interval 240
check_period24x7
notification_period 24x7
notification_optionsc,r
}
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[asterisk-users] Problem with a Zap channel
Hello, I have got two zap channels configured in our asterisk server, one of them is connected to the PSTN directly and the other one is connected to a gsm track, only for mobile calls. Both of them are basic lines. I just connect an iax softphone (idefisk) to the asterisk PBX. If I make a mobile call using the zap channel connected to a gsm track, the mobile I phoned does not hear me nothing. But If the call is made using the zap channel directly connected to the PSTN, both end points hear perfectly. Why this is happening? How could I solve that? Any clue will we wellcomed. Thanks in advance. VoipCrazy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk callerID
Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I make a call directly using one of this line, the callerID is sending correctly. With the same zapata config file and the Freepbx 2.1.3, the callerId was sending correctly. Any clue will be welcome Thanks in advance. VoipCrazy -- zapata.conf-- [channels] language=en context=from-zaptel rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes callerid=asreceived hidecallerid=no callwaiting=yes usecallingpres=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=50 immediate=no rxgain=3.0 txgain=4.0 immediate=no busydetect=yes busycount=8 answeronpolarityswitch=yes hanguponpolarityswitch=yes ringtimeout=8000 faxdetect=both signalling=fxs_ks useincomingcalleridonzaptransfer=yes channel = 1-2 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recomended POE Phones
Hi all, I am looking for phones witch support POE, with a good relation between quality and price to work with asterisk. I just see the Thompson st2030 and the Linksys SPA 922 an SPA 942. Witch of this phones or another ones gave you the best results in a productivity enviroment? Thanks in advance. VoipCrazy. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Activate/Deactivate zap channels in realtime
Hi all, I am looking for a solution for the following problem. I have a little callcenter with 20 agents and 20 incomming analog lines, one for each agent. I need to have abailable as incomming analog lines (FXO Ports) as agents logged, not all the agents are logged all the time. It is needed for example If there are 5 agents logged in asterisk, only would be available 5 analog incomming lines and the rest of the incomming lines asterisk does not answer the call and the call does not arribe to the asterisk machine. Something similar to activate/deactivate the zap channels (one for each incomming analog line), depending the number of agents currently logged in the system. Any clue will be appreciated, VoipCrazy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test to Speech
Hello all, I am looking for software for text to speech in spanish witch works with asterisk (1.2.13). I have tested festival and the cepstral software, both works but the quality is so poor in the spanish language. Someone has worked with any test to speech software with aceptable quality in spanish? Probably in english the text to speech quality will be better. Witch test to speech software gave you the best results in spanish? Thank you VoipCrazy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] snmp Monitor for asterisk boxes
Hello all, Witch snmp system do you use to collect info about their asterisk boxes, for example, uptime, downtime, max load, HD, free memory, asterisk status, ,etc? I have made a look to Cacti and MRTG, but I am not sure they will monitor asterisk. Witch is best snmp system to monitor asterisk based on your experience? Thanks a lot Voip Crazy. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
