Re: [asterisk-users] Async Agi problem
Robert Bielik skrev: Ok, now pretty much everything is up 'n running, however when I try to send an ANSWER (or any) command to *, it replies with org.asteriskjava.manager.response.ManagerError Permission Denied. In manager.conf for the *-java client, I have read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan,agi write = system,call,agent,user,config,command,reporting,originate * is 1.6.1.4 and *-java is 1.0.0 Hmm... setting write = all makes it work... /Rob ___ -- 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] Xorcom device not showing up in /proc
Hello, I remember having a similar problem sometime back but I do not remember the solutions. My Xorcom FXS bank is not showing up in /proc. Here some helpful output: r...@pbx1:~# zaptel_hardware usb:002/006 xpp_usb- e4e4:1161 Astribank-modular USB-firmware pci::01:05.0 wct4xxp+ d161:0210 Wildcard TE210P r...@pbx1:~# ls /proc/zaptel/ 1 2 r...@pbx1:~# ls /proc/bus/usb/ 001 002 devices lsmod: pp 156092 2 xpd_fxs,xpp_usb ftdi_sio 37640 2 usbserial 35688 6 ftdi_sio wmi_acer9644 0 wct4xxp 353920 52 zaptel200068 109 xpd_fxs,xpp,wct4xxp crc_ccitt 3072 1 zaptel button 9232 0 The strange thing is that my configuration worked until I rebooted the server. Best regards, Loïc. ___ -- 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 DID to another server
On Mon, Nov 02, 2009 at 12:19:41PM +0530, DHAVAL INDRODIYA wrote: hello all, i have 2 asterisk boxes on that 1 have public IP Address and another is only have local IP address now on public IP there are some 7 DID forwarded , now i want to forward 3 DID out of 7 DID to local machine we called server B , I know there are DIal , and Switch statement in asterisk , but is there any other convenient way to do this. because if call ratio is high then my call legs become very long. You have just one public IP address? Just one IP connection? If both servers are connected on the same network connection to the internet, the lag wouldn't matter anyway. If server B is on a different connection, the provider can technically forward calls to it once it has been registered. Though this may not work well with the way this provider works, so this may or may not work. So you'll have to provide some more specific details if you want to be more specific. Proper puctuation and proper separation of paragraphs (rather than randomly pressing Enter) will also help making your messages look more readable. -- 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 DID to another server
On Mon, Nov 02, 2009 at 02:49:05AM -0500, ALEX BALASHOV wrote: In a manner of speaking. Top-posting, on top of your other sins. Please spare us this capital punishment. -- 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
[asterisk-users] PSTN Line Parameters Checking
I have Asterisk and Digium AEX808B What are please the commands that i can run on Asterisk to get the information about the connected lines from PSTN to see the parameters of them and as well the corresponding files in Asterisk that i can change into, to tune these parameters to be matched together. Thanks a lot. _ Keep your friends updated—even when you’re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010___ -- 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] Xorcom device not showing up in /proc
On Mon, Nov 02, 2009 at 09:33:21AM +0100, Loic Didelot wrote: Hello, I remember having a similar problem sometime back but I do not remember the solutions. My Xorcom FXS bank is not showing up in /proc. Here some helpful output: r...@pbx1:~# zaptel_hardware usb:002/006 xpp_usb- e4e4:1161 Astribank-modular USB-firmware 1161: the FPGA firmware is not loaded. http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_lsusb_test What version of Zaptel is it? -- 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 DID to another server
Tzafrir Cohen wrote: Top-posting, on top of your other sins. Please spare us this capital punishment. An entirely fair point. Nevertheless, I eagerly await your similarly convicted petitions aimed at curbing illiterate, obnoxious and indolent attempts to get others to do extensive work on one's behalf to fix a problem one has not done the due diligence to rudimentarily understand. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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 DID to another server
Alex, You forgot to clip the extra from the quote, shame on you! On Mon, Nov 2, 2009 at 9:47 AM, Alex Balashov abalas...@evaristesys.com wrote: Tzafrir Cohen wrote: Top-posting, on top of your other sins. Please spare us this capital punishment. An entirely fair point. Nevertheless, I eagerly await your similarly convicted petitions aimed at curbing illiterate, obnoxious and indolent attempts to get others to do extensive work on one's behalf to fix a problem one has not done the due diligence to rudimentarily understand. ___ -- 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] SIP Peers still ping with SIP OPTIONS on a reload
I have the same result with Asterisk 1.4.21 on a Debian Lenny server -- -- -- Marc LEURENT lf...@leurent.eu Le mercredi, 28 octobre 2009 12.27:59, Marc Leurent a écrit : Hello, when I remove a peer from my sip.conf and just do a reload, the peer is still ping with SIP OPTIONS until I restart Asterisk, I use Asterisk 1.4.27-rc2. Is it normal? Thanks As an example, I have added and after removed this lines and ;[sip_trk_vm] ;host=88.191.80.8 ;type=peer ;context=default ;dtmfmode=info ;insecure=port,invite ;nat=never ;sendrpid=yes ;disallow=all ;allow=alaw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension
On Sun, Nov 01, 2009 at 04:13:22PM -0500, cov...@ccs.covici.com wrote: Hi. When I dial a Dahdi extension using asterisk 1.6.0, and there is no answer, the extension hangs up, but the dial status is busy instead of no answer. How do I get this to work -- do I need to update dahdi? The card is an X400p using its FXS module. What version of DAHDI? -- 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] Xorcom device not showing up in /proc
Hi, I am using this version zaptel-1.4.12.9.svn.r4653. Best regards, Loïc. On Mon, 2009-11-02 at 10:38 +0200, Tzafrir Cohen wrote: On Mon, Nov 02, 2009 at 09:33:21AM +0100, Loic Didelot wrote: Hello, I remember having a similar problem sometime back but I do not remember the solutions. My Xorcom FXS bank is not showing up in /proc. Here some helpful output: r...@pbx1:~# zaptel_hardware usb:002/006 xpp_usb- e4e4:1161 Astribank-modular USB-firmware 1161: the FPGA firmware is not loaded. http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_lsusb_test What version of Zaptel is 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] hardware requirements for asterisk
hello friends friend i had just finished my chapters of asterisk. ill be configuring asterisk in for home for r/d purpose. i am having p4 machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the only doubt which i am having is which hardware ill have to buy to configure asterisk. i think analog card ? plz clear my doubt. n be with me from beginning till end, of the journey of asterisk. Regards, Pawan___ -- 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] Xorcom device not showing up in /proc
Hi, can I reset and reload the firmware while the kernel modules are loaded? Should the files appear in /proc once the firmware has been loaded correctly or do I need to unload/reload the kernel modules? Best regards, Loic. On Mon, 2009-11-02 at 10:38 +0200, Tzafrir Cohen wrote: On Mon, Nov 02, 2009 at 09:33:21AM +0100, Loic Didelot wrote: Hello, I remember having a similar problem sometime back but I do not remember the solutions. My Xorcom FXS bank is not showing up in /proc. Here some helpful output: r...@pbx1:~# zaptel_hardware usb:002/006 xpp_usb- e4e4:1161 Astribank-modular USB-firmware 1161: the FPGA firmware is not loaded. http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_lsusb_test What version of Zaptel is 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
Re: [asterisk-users] hardware requirements for asterisk
aster...@opensourcesolution.in wrote: hello friends friend i had just finished my chapters of asterisk. ill be configuring asterisk in for home for r/d purpose. i am having p4 machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the only doubt which i am having is which hardware ill have to buy to configure asterisk. i think analog card ? plz clear my doubt. n be with me from beginning till end, of the journey of asterisk. Depending on what you intend to accomplish, you may not need any additional hardware; you do not need PSTN connectivity to use Asterisk. If you want it anyway, you can get PSTN origination (calls from the PSTN-VoIP) and termination (VoIP-PSTN) over IP without any need for physical lines. If you have a fixed analog line and are determined to interface it with Asterisk, you would need an FXO card. TDM hardware that interfaces with T1/E1 circuits (ISDN PRI, typically) is also available. -- Alex -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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 Fax Module
When we can expect to have a res_fax and res_fax_degium module for asterisk V 1.6.2 Regards Khaled Chehab NGN Eng. Untitled Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com mailto:bs...@mg-tel.com MSN ID :khalidche...@hotmail.com Web Site: http://www.Xplorium.com * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * image001.jpg___ -- 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] hardware requirements for asterisk
On Mon, 2009-11-02 at 09:37 +, aster...@opensourcesolution.in wrote: hello friends friend i had just finished my chapters of asterisk. ill be configuring asterisk in for home for r/d purpose. i am having p4 machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the only doubt which i am having is which hardware ill have to buy to configure asterisk. i think analog card ? plz clear my doubt. n be with me from beginning till end, of the journey of asterisk. Regards, Pawan Hi Pawan, It vey much depend on what you expect the box to be handling As you wrote: soho + RD, i presume it will be anoccasional call. Personally, i would recommend to leave the analogue stuff out of your PC. (no hassle with pci-slots, shared-IRQ's, PSU, ) Leave the handling of analogue-parts to an ATA-box. Linksys (and others) are making those at reasonable prices (Cheaper than an analogue card) hw ___ -- 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] SNOM 870
We have the 870 working great in our test environment so far. Garth van Sittert BSC (Physics Comp Sci) Technical Director BitCo 08600 24826 www.bitco.co.za --[ UxBoD ]-- wrote: Anybody tried one with Asterisk yet ? Views ? Best Regards, ___ -- 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] SNOM 870
- Garth van Sittert ga...@bitco.co.za wrote: | We have the 870 working great in our test environment so far. | | | Garth van Sittert | BSC (Physics Comp Sci) | Technical Director | BitCo | 08600 24826 | www.bitco.co.za | | | | --[ UxBoD ]-- wrote: | Anybody tried one with Asterisk yet ? Views ? | | Best Regards, | | | | How responsive is the touch screen ? Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- 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] Dialstatus
Hi,
you can do print the dialstatus to the console e.g.:
exten = s,n,NoOp(${DIALSTATUS})
More info:
http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp
Bye,
Patrick
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[asterisk-users] MySQL CDR
Hello, Does anyone know where I can get an up to date guide on installing CDR_MSQL? VOIP-Info has old information. Many thanks Dan ___ -- 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] GSM and Wav format
Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these recorded files on WEB. We have following options 1. Record in GSM and send to central Server. Which will convert to it to WAV format using some code / any other thing. The issue in this is that CPU will get very busy in this case. Because GSM Files can be very frequent. 2. Recored in Wav and send to central server. In this case we may face Network Bandwidth problem.(Even we create VPN). QUESTION IS: Is there any other format in which we can record using the record application provided its is small in size and directly playable on WEB. Best Regards Shakeel Abbas ___ -- 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] GSM and Wav format
Hi, at first: why do you use capitals for your name? Don't do that if you don't have a very good reason. You can convert wav to mp3 on the recording server and then send it to the central system. Bye, Patrick On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these recorded files on WEB. We have following options 1. Record in GSM and send to central Server. Which will convert to it to WAV format using some code / any other thing. The issue in this is that CPU will get very busy in this case. Because GSM Files can be very frequent. 2. Recored in Wav and send to central server. In this case we may face Network Bandwidth problem.(Even we create VPN). QUESTION IS: Is there any other format in which we can record using the record application provided its is small in size and directly playable on WEB. Best Regards Shakeel Abbas ___ -- 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 -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Tel. +49 2151 5554-263 Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- 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] supermicro hardware + sangoma
hi, i have new supermicro server (centos5, 2.6.30.9/2.6.27.34/2.6.18-distro kernels, wanpipe 3.5.6) card is: 1 . AFT-A101-SH : SLOT=4 : BUS=8 : IRQ=11 : CPU=A : PORT=1 : HWEC=32 : V=36 and i have this in log irq 17: nobody cared (try booting with the irqpoll option) Pid: 0, comm: swapper Not tainted 2.6.30.9rh #1 Call Trace: [c0465b07] __report_bad_irq+0x27/0x90 [c0465caa] note_interrupt+0x13a/0x180 [c04665af] handle_fasteoi_irq+0x9f/0xd0 [c0466510] ? handle_fasteoi_irq+0x0/0xd0 IRQ [c0404506] ? do_IRQ+0x46/0xb0 [c0588234] ? acpi_hw_write_port+0x27/0x71 [c0403469] ? common_interrupt+0x29/0x30 [c05943d4] ? acpi_idle_enter_bm+0x218/0x241 [c062bf8e] ? cpuidle_idle_call+0x6e/0xc0 [c0401e45] ? cpu_idle+0x35/0x60 [c06d42f2] ? start_secondary+0x182/0x1e0 handlers: [f872d9b0] (sdla_isr+0x0/0x310 [wanpipe]) Disabling IRQ #17 dou you have idea what is the problem? irqpoll doesnt help i have tried this supermicro motherboards http://www.supermicro.com/products/motherboard/QPI/5500/X8DTU-F.cfm http://www.supermicro.com/products/motherboard/QPI/5500/X8DTi-F.cfm http://www.supermicro.com/products/motherboard/QPI/5500/X8DTL-iF.cfm do you have someone working sangoma card with Tylersburg(intel 5520/5500) chipset? thanks p.s. sorry for offtopic :( --- Marek Cervenka === ___ -- 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] GSM and Wav format
Thanks Patrick. First: I dont do that intentionally. Thanks for suggestion. Let me investigate it. On Mon, Nov 2, 2009 at 5:34 PM, Patrick Plattes patr...@erdbeere.netwrote: Hi, at first: why do you use capitals for your name? Don't do that if you don't have a very good reason. You can convert wav to mp3 on the recording server and then send it to the central system. Bye, Patrick On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these recorded files on WEB. We have following options 1. Record in GSM and send to central Server. Which will convert to it to WAV format using some code / any other thing. The issue in this is that CPU will get very busy in this case. Because GSM Files can be very frequent. 2. Recored in Wav and send to central server. In this case we may face Network Bandwidth problem.(Even we create VPN). QUESTION IS: Is there any other format in which we can record using the record application provided its is small in size and directly playable on WEB. Best Regards Shakeel Abbas ___ -- 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 -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Tel. +49 2151 5554-263 Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- 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] GSM and Wav format
After conversion from .wav to .mp3 the size remains almost the same. On Mon, Nov 2, 2009 at 5:46 PM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: Thanks Patrick. First: I dont do that intentionally. Thanks for suggestion. Let me investigate it. On Mon, Nov 2, 2009 at 5:34 PM, Patrick Plattes patr...@erdbeere.netwrote: Hi, at first: why do you use capitals for your name? Don't do that if you don't have a very good reason. You can convert wav to mp3 on the recording server and then send it to the central system. Bye, Patrick On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these recorded files on WEB. We have following options 1. Record in GSM and send to central Server. Which will convert to it to WAV format using some code / any other thing. The issue in this is that CPU will get very busy in this case. Because GSM Files can be very frequent. 2. Recored in Wav and send to central server. In this case we may face Network Bandwidth problem.(Even we create VPN). QUESTION IS: Is there any other format in which we can record using the record application provided its is small in size and directly playable on WEB. Best Regards Shakeel Abbas ___ -- 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 -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Tel. +49 2151 5554-263 Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas ___ -- 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] Xorcom device not showing up in /proc
On Mon, Nov 02, 2009 at 10:38:45AM +0100, Loic Didelot wrote: Hi, can I reset and reload the firmware while the kernel modules are loaded? Yes, there shouldn't be a problem with that. Should the files appear in /proc once the firmware has been loaded correctly or do I need to unload/reload the kernel modules? Kernel loading is purely userspace. -- 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] GSM and Wav format
On Mon, Nov 02, 2009 at 05:11:53PM +0500, ABBAS SHAKEEL wrote: Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that .WAV (wav49, wav/gsm) should be playable by most systems. -- 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] Xorcom device not showing up in /proc
Hello, I now have this which looks better. r...@pbx1:~# lsusb Bus 002 Device 010: ID e4e4:1162 Bus 002 Device 001: ID : Bus 001 Device 004: ID 0403:e6c8 Future Technology Devices Bus 001 Device 003: ID 0403:6001 Future Technology Devices Bus 001 Device 001: ID : r...@pbx1:~# zaptel_hardware usb:002/010 xpp_usb- e4e4:1162 Astribank-modular FPGA-firmware pci::01:05.0 wct4xxp+ d161:0210 Wildcard TE210P But the device is still not listed in /proc Best regards, Loïc. On Mon, 2009-11-02 at 15:10 +0200, Tzafrir Cohen wrote: On Mon, Nov 02, 2009 at 10:38:45AM +0100, Loic Didelot wrote: Hi, can I reset and reload the firmware while the kernel modules are loaded? Yes, there shouldn't be a problem with that. Should the files appear in /proc once the firmware has been loaded correctly or do I need to unload/reload the kernel modules? Kernel loading is purely userspace. -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 ldide...@mixvoip.com http://www.mixvoip.com ___ -- 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] Xorcom device not showing up in /proc
On Mon, Nov 02, 2009 at 02:28:36PM +0100, Loic Didelot wrote: Hello, I now have this which looks better. r...@pbx1:~# lsusb Bus 002 Device 010: ID e4e4:1162 Bus 002 Device 001: ID : Bus 001 Device 004: ID 0403:e6c8 Future Technology Devices Bus 001 Device 003: ID 0403:6001 Future Technology Devices Bus 001 Device 001: ID : r...@pbx1:~# zaptel_hardware usb:002/010 xpp_usb- e4e4:1162 Astribank-modular FPGA-firmware pci::01:05.0 wct4xxp+ d161:0210 Wildcard TE210P That is odd. Now is the time to look at /var/log/kern.log . -- 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] asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension
svn 6466 from trunk. Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Nov 01, 2009 at 04:13:22PM -0500, cov...@ccs.covici.com wrote: Hi. When I dial a Dahdi extension using asterisk 1.6.0, and there is no answer, the extension hangs up, but the dial status is busy instead of no answer. How do I get this to work -- do I need to update dahdi? The card is an X400p using its FXS module. What version of DAHDI? -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- 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] pattern matching DID
On Sun, 2009-11-01 at 18:50 -0500, Thomas Perron wrote: Where is everyone located? I am in Virginia, USA There are literally thousands of people on this mailing list, so I doubt it's worth having everyone tell you where they're from. That being said, I'm also in Virginia (near Fredericksburg), and there's enough interest in the area that we might start up a local Asterisk users group in the area. What part of Virginia are you from? -- Jared Smith Training Manager Digium, 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] GSM and Wav format
FWIW, I convert all of my files to WAV for Web reading using SOX. Sox will let you put all your files into the compressed gsm format for storage (sox file.wav file.gsm), then you can just reverse the process for presentation (sox file.gsm /tmp/file.wav) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ABBAS SHAKEEL Sent: Monday, November 02, 2009 6:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] GSM and Wav format After conversion from .wav to .mp3 the size remains almost the same. On Mon, Nov 2, 2009 at 5:46 PM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Thanks Patrick. First: I dont do that intentionally. Thanks for suggestion. Let me investigate it. On Mon, Nov 2, 2009 at 5:34 PM, Patrick Plattes patr...@erdbeere.net wrote: Hi, at first: why do you use capitals for your name? Don't do that if you don't have a very good reason. You can convert wav to mp3 on the recording server and then send it to the central system. Bye, Patrick On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these recorded files on WEB. We have following options 1. Record in GSM and send to central Server. Which will convert to it to WAV format using some code / any other thing. The issue in this is that CPU will get very busy in this case. Because GSM Files can be very frequent. 2. Recored in Wav and send to central server. In this case we may face Network Bandwidth problem.(Even we create VPN). QUESTION IS: Is there any other format in which we can record using the record application provided its is small in size and directly playable on WEB. Best Regards Shakeel Abbas ___ -- 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 -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Tel. +49 2151 5554-263 Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas ___ -- 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] IVR
As I understand this thread, you want two different contexts based on the number you dial. If you dial 1703... the big10 context should be executed. If 1567... then Cleveland is executed. Is this correct? If so Then this is what the two lines in [default] should read: exten = _1703.,1,Goto(big10-IVR,s,1) exten = _1567.,1,Goto(cleveland-IVR,s,1) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Samuel Nair Sent: Monday, November 02, 2009 12:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR Try running your asterisk service with the -vvvc option or connect to it via the -r option, and then try making a call that would cause it to land in the default context, you will see the way asterisk traverses the dial plan, this will give you good debug info. sam!! Thomas Perron wrote: Hi Juan, I have this: [default] ;include = stdexten include = big10-IVR include = cleveland-IVR exten = _1703XXX,1,Goto(big10-IVR,s,1) exten = _1567XXX,1,Goto(cleveland-IVR,s,1) You recommend I have this: [default] exten = _1703XXX,1,Goto(big10-IVR,s,1) exten = _1567XXX,1,Goto(cleveland-IVR,s,1) I tried this and it does not seem to work. Other thoughts? Where located please? 2009/11/1 Juan E. Rodríguez jerdg...@gmail.com mailto:jerdg...@gmail.com As I see here, you do not have to include the big10 context inside the default context, as you have an extension defined to reach that context and its extention is start extension. If the cleveland-IVR is based on the start extension too, the same applies. Besides that, it would work...(maybe not the way you expect... :-) ) Regards, Juan Thomas Perron wrote: Is this going to work: [default] include = stdexten include = big10-IVR include = cleveland-IVR exten = _17035745353,1,Goto(big10-IVR,s,1) exten = _15672528431,1,Goto(cleveland-IVR,s,1) [big10-IVR] exten = s,1,Answer() exten = s,n,Background(dir-welcome) ;exten = s,n,WaitExten(1) ;exten = s,n,Background(astcc-please-enter-your) ___ -- 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 ___ -- 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] change L(x[:y][:z]) parameter of DIAL command after call is bridged
Hi, is there any way from outside change x,y an z after a call is bridged? maybe with AMI interface? best regards Thomas ___ -- 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] Remote Party ID
Hello, Does anyone know how to set the remote party id? Thanks Dan Journo ___ -- 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] Dialstatus
On Mon, 2 Nov 2009, Patrick Plattes wrote:
you can do print the dialstatus to the console e.g.:
exten = s,n,NoOp(${DIALSTATUS})
More info:
http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp
A better practice would be to use verbose() -- an application with
greater functionality written specifically for this purpose.
--
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] Async Agi problem
because read=agi lets you read agi events, not send agi actions, agi in write= must be set too if you want to send agi commands. On Mon, Nov 2, 2009 at 3:25 AM, Robert Bielik robert.bie...@xponaut.sewrote: Robert Bielik skrev: Ok, now pretty much everything is up 'n running, however when I try to send an ANSWER (or any) command to *, it replies with org.asteriskjava.manager.response.ManagerError Permission Denied. In manager.conf for the *-java client, I have read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan,agi write = system,call,agent,user,config,command,reporting,originate * is 1.6.1.4 and *-java is 1.0.0 Hmm... setting write = all makes it work... /Rob ___ -- 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 -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- 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] Remote IP Phone's
Hi all, I am wondering what people are doing for security when registering IP phone's remotely if you do not have the equipment to do a VPN tunnel at the remote site. The phone I would be working with mainly is the Polycom lineup. Thanks, Connor Spiess ___ -- 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] Dynamic DNS trunk --- SOLVED
dnsmgr.conf: enable=yes refreshinterval=300 regards. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 30, 2009 3:28 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Dynamic DNS trunk Thanks I did this dnsmgr.conf: enable=yes refreshinterval=300 I did dnsmgr refresh, the DNS in the trunk did not got the new ip, also I waited 5 min. do I have to add an entry to dnsmgr?? Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Friday, October 30, 2009 1:53 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dynamic DNS trunk On 30/10/09 6:42 AM, B.Masoud @ SH wrote: Hi I tried with registration, it did not update the IP address I can only see it updated if I typed: Sip reload I have few questions: Is there any way Asterisk automatically updates the DNS? Yep /etc/asterisk/dnsmgr.conf -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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
Re: [asterisk-users] Remote IP Phone's
You can restrict on IP address, MAC address and port type and that's just what I know. If someone want's through bad enough you're going to have a problem, but you can at least slow down or stop casual hackers. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Connor Spiess Sent: Monday, November 02, 2009 9:38 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Remote IP Phone's Hi all, I am wondering what people are doing for security when registering IP phone's remotely if you do not have the equipment to do a VPN tunnel at the remote site. The phone I would be working with mainly is the Polycom lineup. Thanks, Connor Spiess ___ -- 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] supermicro hardware + sangoma
Hi, Try Berofix / beronet - tested with Tylersburg Supermicro mb's - works perfectly well. Jacek -Original Message-From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenkaSent: Monday, November 02, 2009 1:38 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] supermicro hardware + sangoma hi, i have new supermicro server (centos5, 2.6.30.9/2.6.27.34/2.6.18-distro kernels, wanpipe 3.5.6) card is: 1 . AFT-A101-SH : SLOT=4 : BUS=8 : IRQ=11 : CPU=A : PORT=1 : HWEC=32 : V=36 and i have this in log irq 17: nobody cared (try booting with the "irqpoll" option) Pid: 0, comm: swapper Not tainted 2.6.30.9rh #1 Call Trace: [c0465b07] __report_bad_irq+0x27/0x90 [c0465caa] note_interrupt+0x13a/0x180 [c04665af] handle_fasteoi_irq+0x9f/0xd0 [c0466510] ? handle_fasteoi_irq+0x0/0xd0 IRQ [c0404506] ? do_IRQ+0x46/0xb0 [c0588234] ? acpi_hw_write_port+0x27/0x71 [c0403469] ? common_interrupt+0x29/0x30 [c05943d4] ? acpi_idle_enter_bm+0x218/0x241 [c062bf8e] ? cpuidle_idle_call+0x6e/0xc0 [c0401e45] ? cpu_idle+0x35/0x60 [c06d42f2] ? start_secondary+0x182/0x1e0 handlers: [f872d9b0] (sdla_isr+0x0/0x310 [wanpipe]) Disabling IRQ #17 dou you have idea what is the problem? irqpoll doesnt help i have tried this supermicro motherboards http://www.supermicro.com/products/motherboard/QPI/5500/X8DTU-F.cfm http://www.supermicro.com/products/motherboard/QPI/5500/X8DTi-F.cfm http://www.supermicro.com/products/motherboard/QPI/5500/X8DTL-iF.cfm do you have someone working sangoma card with Tylersburg(intel 5520/5500) chipset? thanks p.s. sorry for offtopic :( --- Marek Cervenka === ___ -- 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] Remote IP Phone's
Hello Connor, You might be able to start with this link: http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret You can also go even further if you know the IP address of where the phones are coming from by using the permit/deny options: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+permit-deny-mask Hope that helps. Dean Hoover Connor Spiess wrote: Hi all, I am wondering what people are doing for security when registering IP phone’s remotely if you do not have the equipment to do a VPN tunnel at the remote site. The phone I would be working with mainly is the Polycom lineup. Thanks, Connor Spiess ___ -- 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] SNOM 870
On Fri, 30 Oct 2009, hbk wrote: Hi, I have played with the 820 for some weeks, mostly love it excellent speech quality. Even got the mini browser running showing my favorite webcam, this could be put to real use too:) Issues so far: Some embarrassing crashes while speaking, was able to speak but all freezed. Still a little fresh firmware I guess. Error 404 after showing webcam picture, but it works! Have to use *1 to start recording, record soft button does not seem to work with *. Still I recommend it, best IP phone I have tried! Not sure 870 is worth the extra money, not tested that yet. How is the build quality of the 870? The mortality rate on power supplies, diplays and the number or broken receiver hook swicthes on the lot of Snom 360's i bought 3 years ago is outright embarrassing. ___ -- 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 Fax Module
Probably after 1.6.2 has been officially released beyond the release candidate stage. Thanks, --Warren Selby On Mon, Nov 2, 2009 at 4:14 AM, Khaled W Chehab kche...@xplorium.comwrote: When we can expect to have a res_fax and res_fax_degium module for asterisk V 1.6.2 Regards *Khaled Chehab* * NGN Eng.* [image: Untitled] * Operations Office - Lebanon* Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com bs...@mg-tel.com MSN ID :khalidche...@hotmail.com Web Site: http://www.Xplorium.com -- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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 image001.jpg___ -- 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] MySQL CDR
What version of asterisk are you installing? Thanks, --Warren Selby On Mon, Nov 2, 2009 at 5:59 AM, Dan Journo d...@keshercommunications.comwrote: Hello, Does anyone know where I can get an up to date guide on installing CDR_MSQL? VOIP-Info has old information. Many thanks Dan ___ -- 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] Asterisk Fax Module
Just my .02; You shouldn't use outlying features like fax on rc releases - these aren't usually but can be (b)leading edge stuff. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Monday, November 02, 2009 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Fax Module Probably after 1.6.2 has been officially released beyond the release candidate stage. Thanks, --Warren Selby On Mon, Nov 2, 2009 at 4:14 AM, Khaled W Chehab kche...@xplorium.com wrote: When we can expect to have a res_fax and res_fax_degium module for asterisk V 1.6.2 Regards Khaled Chehab NGN Eng. Untitled Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com mailto:bs...@mg-tel.com MSN ID :khalidche...@hotmail.com Web Site: http://www.Xplorium.com _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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 image001.jpg___ -- 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] Skyp SIP? - what is free for a home *
There is something called Opensky that claims to allow making and receiving skype calls on your SIP device (including Asterisk). However I haven´t tested it yet. http://latestgeeknews.blogspot.com/2009/02/opensky-skype-interface-gateway-who.html 2009/11/1 hbk fo...@online.no Hi, I get confused about all solutions for Skype! I want to connect as simple as possible out home * to be able to at least answer Skype calls. Now I use a PC USB box and a FXO, works ok both call directions but uses a PC. Any good and free idea ? Thank you! HB ___ -- 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] SNOM 870
Remco Barendse wrote: On Fri, 30 Oct 2009, hbk wrote: Hi, I have played with the 820 for some weeks, mostly love it excellent speech quality. Even got the mini browser running showing my favorite webcam, this could be put to real use too:) Issues so far: Some embarrassing crashes while speaking, was able to speak but all freezed. Still a little fresh firmware I guess. Error 404 after showing webcam picture, but it works! Have to use *1 to start recording, record soft button does not seem to work with *. Still I recommend it, best IP phone I have tried! Not sure 870 is worth the extra money, not tested that yet. How is the build quality of the 870? The mortality rate on power supplies, diplays and the number or broken receiver hook swicthes on the lot of Snom 360's i bought 3 years ago is outright embarrassing. That's odd. We've had Snom 190s, 320s, and 360s running day in day out for years with not a single issue. Maybe we got all the good ones from your batch. If that's the case, I thank you for 'taking one for the team' as it were. ;) N. ___ -- 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] SNOM 870
The mortality rate on power supplies, diplays and the number or broken receiver hook swicthes on the lot of Snom 360's i bought 3 years ago is outright embarrassing. That's odd. We've had Snom 190s, 320s, and 360s running day in day out for years with not a single issue. Would say about 1 in 10 PSUs fail in 2 years. About 1 in 20 in 2 years for screens. Never had a hook switch fail. 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] DAHDI/ZAP overlap dialing
--- On Sat, 10/31/09, Martin asteriskl...@callthem.info wrote: On Sat, Oct 31, 2009 at 5:27 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: I'm not sure if handling of overlap hasn't changed since. But can you provide a trace of how Asterisk sees things? e.g. 'pri intense debug span 1' the intense debug is overkill we only need messages of layer 3 ... just do pri debug span 1 Martin Here's the pri trace: Nov 2 17:22:28 VERBOSE[11329] logger.c: Protocol Discriminator: Q.931 (8) len=38 Nov 2 17:22:28 VERBOSE[11329] logger.c: Call Ref: len= 2 (reference 16976/0x4250) (Originator) Nov 2 17:22:28 VERBOSE[11329] logger.c: Message type: SETUP (5) Nov 2 17:22:28 VERBOSE[11329] logger.c: [04 03 80 90 a3] Nov 2 17:22:28 VERBOSE[11329] logger.c: Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Nov 2 17:22:28 VERBOSE[11329] logger.c: Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Nov 2 17:22:28 VERBOSE[11329] logger.c: User information layer 1: A-Law (35) Nov 2 17:22:28 VERBOSE[11329] logger.c: [18 03 a9 83 8b] Nov 2 17:22:28 VERBOSE[11329] logger.c: Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 Nov 2 17:22:28 VERBOSE[11329] logger.c: ChanSel: As indicated in following octets Nov 2 17:22:28 VERBOSE[11329] logger.c:Ext: 1 Coding: 0 Number Specified Channel Type: 3 Nov 2 17:22:28 VERBOSE[11329] logger.c:Ext: 1 Channel: 11 ] Nov 2 17:22:28 VERBOSE[11329] logger.c: [1e 02 80 83] Nov 2 17:22:28 VERBOSE[11329] logger.c: Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Nov 2 17:22:28 VERBOSE[11329] logger.c:Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] Nov 2 17:22:28 VERBOSE[11329] logger.c: [6c 06 00 81 37 30 33 34] Nov 2 17:22:28 VERBOSE[11329] logger.c: Calling Number (len= 8) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Nov 2 17:22:28 VERBOSE[11329] logger.c: Presentation: Presentation permitted, user number passed network screening (1) '7034' ] Nov 2 17:22:28 VERBOSE[11329] logger.c: [70 05 80 31 30 30 34] Nov 2 17:22:28 VERBOSE[11329] logger.c: Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '1004' ] Nov 2 17:22:28 VERBOSE[11329] logger.c: [7d 02 91 81] Nov 2 17:22:28 VERBOSE[11329] logger.c: IE: High-layer Compatibility (len = 4) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Making new call for cr 16976 Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing Q.931 Call Setup Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 4 (cs0, Bearer Capability) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 24 (cs0, Channel Identification) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 30 (cs0, Progress Indicator) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 108 (cs0, Calling Party Number) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 112 (cs0, Called Party Number) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 125 (cs0, High-layer Compatibility) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Extension '1004' in context 'from-pstn-deviate-custom' from '7034' does not exist. Rejecting call on channel 1/11, span 1 Nov 2 17:22:28 VERBOSE[11329] logger.c: NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Nov 2 17:22:28 VERBOSE[11329] logger.c: Protocol Discriminator: Q.931 (8) len=9 Nov 2 17:22:28 VERBOSE[11329] logger.c: Call Ref: len= 2 (reference 16976/0x4250) (Terminator) Nov 2 17:22:28 VERBOSE[11329] logger.c: Message type: RELEASE COMPLETE (90) Nov 2 17:22:28 VERBOSE[11329] logger.c: [08 02 81 81] Nov 2 17:22:28 VERBOSE[11329] logger.c: Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Nov 2 17:22:28 VERBOSE[11329] logger.c: Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] Nov 2 17:22:28 VERBOSE[11329] logger.c: NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null Nov 2 17:22:28 VERBOSE[11329] logger.c: NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null The 'from-pstn-deviate-custom' context has lines such as: exten = _100[14567]XXX,1,... exten = _100[14567]XXX,n,... Any ideas? Vieri ___ -- 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] Nagios check_asterisk_peers needs rights to question the Asterisk-server
When executing the following command : [r...@nagios ~]# /usr/local/nagios/libexec/check_nrpe -H ip_address -c check_asterisk_peers I get the following output : NRPE: Unable to read output Somewhere Nagios does not have enough rights to question Asterisk about the sip peers. These are the rights of the check_nrpe on the Nagios Server : [r...@nagios ~]# ls -l /usr/local/nagios/libexec/check_nrpe -rwxrwxr-x 1 nagios nagios 58017 Oct 31 11:40 /usr/local/nagios/libexec/check_nrpe These are the rights of the plugin that questions about the SIP-peers on the Asterisk-server : bash-3.2# ls -l /usr/local/nagios/libexec/nagisk.pl -rwxr-x--- 1 nagios nagios 4163 Nov 2 17:12 /usr/local/nagios/libexec/nagisk.pl The NRPE-plugin on the Asterisk-server is part of the Xinetd-proces. Asterisk himself is currently running as the root-user. Question : I'm confused about which proces/plugin I need to give more rights so that the Asterisk-server can be questioned about its sip-peers. Jonas. ___ -- 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] Dialstatus
On 11/02/09 07:28, Steve Edwards wrote:
On Mon, 2 Nov 2009, Patrick Plattes wrote:
you can do print the dialstatus to the console e.g.:
exten = s,n,NoOp(${DIALSTATUS})
More info:
http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp
A better practice would be to use verbose() -- an application with
greater functionality written specifically for this purpose.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
What was looking to do something in macro after the channel gets connected
dialstatus=Answer but it doesn't work.
Running the macro I don't hear anything (only a dial tone) until the macro is
finished.
--
Joseph
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Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6
We were thinking about doing something similar as well. A lot of people are asking for this. If there is anybody else interested, we could share the load I was thinking about creating a context like @agents, so that when you do the log-on you basically add Local/1...@agents as a member of the queue. When you ring it, it basically looks up for the actual device in AstDB and dials it like: Queue - (member) Local/1...@agents - (astdb) SIP/234 I think that we should be able to forward channel state as well (using hints? I've never done it) so that app_queue does not try dialling agents that are busy. I was thinking about storing queue-agent associations into config strings, and/or AstDB, and/or http over curl. And yes, ideally it should work fine on 1.4's as well Things that should be working from version one: - logging compatible with older asterisk's - authentication using Voicemail -.plug and play on most systems - channel states - pause/unpause with pause codes - ...you tell me Anybody interested? l. 2009/10/30 Mariano Lecuona mlecu...@gmail.com Hi all, I would like to know if there is any application replacement for the AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from what I've read that the call back login agent can be done using a smart dialplan as showed on the docs. But I cannot thinks a flexible dialplan for a dinamic reassignation of agents to different queues every day. Thanks in advance. Mariano ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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] Real replacement for AgentCallBackLogin() on Asterisk 1.6
To avoid boring everybody else to death with the discussion, I created a mailing list for that on Google Groups - see http://tinyurl.com/yjtf62s Thanks l. 2009/11/2 Lenz Emilitri lenz.lo...@gmail.com We were thinking about doing something similar as well. A lot of people are asking for this. If there is anybody else interested, we could share the load I was thinking about creating a context like @agents, so that when you do the log-on you basically add Local/1...@agents as a member of the queue. When you ring it, it basically looks up for the actual device in AstDB and dials it like: Queue - (member) Local/1...@agents - (astdb) SIP/234 I think that we should be able to forward channel state as well (using hints? I've never done it) so that app_queue does not try dialling agents that are busy. I was thinking about storing queue-agent associations into config strings, and/or AstDB, and/or http over curl. And yes, ideally it should work fine on 1.4's as well Things that should be working from version one: - logging compatible with older asterisk's - authentication using Voicemail -.plug and play on most systems - channel states - pause/unpause with pause codes - ...you tell me Anybody interested? l. 2009/10/30 Mariano Lecuona mlecu...@gmail.com Hi all, I would like to know if there is any application replacement for the AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from what I've read that the call back login agent can be done using a smart dialplan as showed on the docs. But I cannot thinks a flexible dialplan for a dinamic reassignation of agents to different queues every day. Thanks in advance. Mariano ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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] Real replacement for AgentCallBackLogin() on Asterisk 1.6
I have sent this before but here is how I do agent login and queue:
; Agent login logout
exten = *20,1,Verbose(2,Doing agent login/logout)
exten = *20,n,Answer()
exten = *20,n,wait(.0.5)
exten = *20,n,Read(AgentNumber,agent-user)
exten = *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten = *20,n,GotoIf($[${UserID}=]?NOUSER)
exten = *20,n,Set(AgentStatus=${DB(users/${UserID}/AgentStatus)})
exten = *20,n,GotoIf($[${AgentStatus}=1]?VERIFY)
exten = *20,n,GotoIf($[${AgentStatus}=2]?VERIFY)
exten = *20,n(NOUSER),Playback(cfmc/bad-agent)
exten = *20,n,Playback(vm-goodbye)
exten = *20,n,Hangup()
exten = *20,n(VERIFY),VMAuthenticate(${agentnumb...@ourvm)
exten = *20,n,GotoIf($[${AgentStatus}=2]?AGENTOFF)
exten = *20,n,Set(DB(users/${UserID}/AgentStatus)=2)
exten = *20,n,Set(DB(users/${UserID}/AgentDevice)=${CUT(CHANNEL,-,1)})
exten = *20,n,AddQueueMember(support,Local/Queue${AgentNumber}
@ansqueue${C$
; AQMSTATUS can be ADDED | MEMBERALREADY | NOSUCHQUEUE
exten = *20,n,Playback(agent-loginok)
exten = *20,n,Verbose(2,Agent ${AgentNumber} added ${DB(users/$
{UserID}/AgentD$
exten = *20,n,Hangup()
exten = *20,n(AGENTOFF),Set(DB(users/${UserID}/AgentStatus)=1)
exten = *20,n,Set(OldVal=${DB_DELETE(users/${UserID}/AgentDevice)})
exten = *20,n,RemoveQueueMember(support,Local/Queue${AgentNumber}
@ansqueue)
exten = *20,n,Playback(agent-loggedoff)
exten = *20,n,Verbose(2,Agent ${AgentNumber} removed)
exten = *20,n,Hangup()
exten = 201,1,Verbose(2,Doing support call)
exten = 201,n,Answer()
exten = 201,n,Wait(0.5)
;exten = 201,n,Set(qac=${QUEUE_MEMBER(support,free)})
exten = 201,n,Set(qac=${QUEUE_MEMBER_COUNT(support)})
exten = 201,n,GotoIf($[${qac} 0]?HAVEAGNT)
exten = 201,n,Verbose(2,No agents free in support queue)
exten = 201,n,Playback(cfmc/support-no-agent)
exten = 201,n,Voicemail(2...@ourvm,u)
exten = 201,n,Playback(goodbye)
exten = 201,n,Hangup()
exten = 201,n(HAVEAGNT),Playback(cfmc/support-intro)
exten = 201,n,Verbose(2,Queuing caller for support agent)
exten = 201,n,Queue(support,nrt,,,120)
exten = 201,n,Verbose(2,Support agent did not answer call)
exten = 201,n,Voicemail(2...@ourvm,b)
exten = 201,n,Playback(goodbye)
exten = 201,n,Hangup()
[ansqueue]
exten = _Queue.,1,Set(AgentNumber=${EXTEN:5})
exten = _Queue.,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten = _Queue.,n,Set(AgentDevice=${DB(users/${UserID}/AgentDevice)})
;exten = _Queue.,n,Verbose(2,Agent ${AgentNumber} status is $
{DEVICE_STATE(${A$
exten = _Queue.,n,Verbose(2,Agent ${AgentNumber} status is ${DEVSTATE
(${AgentD$
;exten = _Queue.,n,GotoIf($[${DEVICE_STATE($
{AgentDevice})}=NOT_INUSE]?DIA$
exten = _Queue.,n,GotoIf($[${DEVSTATE(${AgentDevice})}=NOT_INUSE]?
DIALIT)
exten = _Queue.,n,Busy()
exten = _Queue.,n,Hangup()
exten = _Queue.,n(DIALIT),Dial(${AgentDevice},,g)
exten = _Queue.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 2, 2009, at 8:44 AM, Lenz Emilitri wrote:
We were thinking about doing something similar as well. A lot of
people are asking for this. If there is anybody else interested, we
could share the load
I was thinking about creating a context like @agents, so that when
you do the log-on you basically add Local/1...@agents as a member of
the queue. When you ring it, it basically looks up for the actual
device in AstDB and dials it like:
Queue - (member) Local/1...@agents - (astdb) SIP/234
I think that we should be able to forward channel state as well
(using hints? I've never done it) so that app_queue does not try
dialling agents that are busy.
I was thinking about storing queue-agent associations into config
strings, and/or AstDB, and/or http over curl. And yes, ideally it
should work fine on 1.4's as well
Things that should be working from version one:
- logging compatible with older asterisk's
- authentication using Voicemail
-.plug and play on most systems
- channel states
- pause/unpause with pause codes
- ...you tell me
Anybody interested?
l.
2009/10/30 Mariano Lecuona mlecu...@gmail.com
Hi all,
I would like to know if there is any application replacement for the
AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from
what I've read that the call back login agent can be done using a
smart dialplan as showed on the docs. But I cannot thinks a flexible
dialplan for a dinamic reassignation of agents to different queues
every day.
Thanks in advance.
Mariano
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[asterisk-users] Asterisk 1.4 and Fax
Hi, Does anyone have an up to date guide for setting up fax 2 email with asterisk? Thanks Dan IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from £150 per month, please contact Kesher Communications. Dan Journo IT Business Consultant Kesher Communications Ltd Tel: 07957 233 599 Web: http://www.KesherCommunications.com http://www.keshercommunications.com/ Live Chat/Instant Support: Click Here http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7F508C6B61A91700343lang=ensurpre=PreSurvey This email and any files transmitted with it are confidential and intended solely for the recipient(s). If you are not the named addressee you should not disseminate, copy or alter this email. Under no circumstances may this email be distributed without written permission from the sender. Warning: Although the Company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. All prices exclude VAT unless otherwise stated. No responsibility is taken for any recommendations made by a sender or by Kesher Communications Ltd. Recipient(s) takes responsibility for any actions taken as a result of advice and recommendations given by Kesher Communications Ltd. image001.jpgimage002.gif___ -- 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] DAHDI/ZAP overlap dialing
I can only tell you that it worked before... Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Extension '1004' in context 'from-pstn-deviate-custom' from '7034' does not when you have overlapdial turned on it should have checked if there's a potential matching extension which you have it right there and asterisk should have sent SETUP_ACK message back. if you won't find the solution for this I might fix that as a bounty if you're interested I'd double check that you really have overlapdial=yes for those channels ... it should be declared before channel = keyword in zapata.conf/chan_dahdi.conf Martin On Mon, Nov 2, 2009 at 10:28 AM, Vieri rentor...@yahoo.com wrote: --- On Sat, 10/31/09, Martin asteriskl...@callthem.info wrote: On Sat, Oct 31, 2009 at 5:27 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: I'm not sure if handling of overlap hasn't changed since. But can you provide a trace of how Asterisk sees things? e.g. 'pri intense debug span 1' the intense debug is overkill we only need messages of layer 3 ... just do pri debug span 1 Martin Here's the pri trace: Nov 2 17:22:28 VERBOSE[11329] logger.c: Protocol Discriminator: Q.931 (8) len=38 Nov 2 17:22:28 VERBOSE[11329] logger.c: Call Ref: len= 2 (reference 16976/0x4250) (Originator) Nov 2 17:22:28 VERBOSE[11329] logger.c: Message type: SETUP (5) Nov 2 17:22:28 VERBOSE[11329] logger.c: [04 03 80 90 a3] Nov 2 17:22:28 VERBOSE[11329] logger.c: Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Nov 2 17:22:28 VERBOSE[11329] logger.c: Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Nov 2 17:22:28 VERBOSE[11329] logger.c: User information layer 1: A-Law (35) Nov 2 17:22:28 VERBOSE[11329] logger.c: [18 03 a9 83 8b] Nov 2 17:22:28 VERBOSE[11329] logger.c: Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 Nov 2 17:22:28 VERBOSE[11329] logger.c: ChanSel: As indicated in following octets Nov 2 17:22:28 VERBOSE[11329] logger.c: Ext: 1 Coding: 0 Number Specified Channel Type: 3 Nov 2 17:22:28 VERBOSE[11329] logger.c: Ext: 1 Channel: 11 ] Nov 2 17:22:28 VERBOSE[11329] logger.c: [1e 02 80 83] Nov 2 17:22:28 VERBOSE[11329] logger.c: Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Nov 2 17:22:28 VERBOSE[11329] logger.c: Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] Nov 2 17:22:28 VERBOSE[11329] logger.c: [6c 06 00 81 37 30 33 34] Nov 2 17:22:28 VERBOSE[11329] logger.c: Calling Number (len= 8) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Nov 2 17:22:28 VERBOSE[11329] logger.c: Presentation: Presentation permitted, user number passed network screening (1) '7034' ] Nov 2 17:22:28 VERBOSE[11329] logger.c: [70 05 80 31 30 30 34] Nov 2 17:22:28 VERBOSE[11329] logger.c: Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '1004' ] Nov 2 17:22:28 VERBOSE[11329] logger.c: [7d 02 91 81] Nov 2 17:22:28 VERBOSE[11329] logger.c: IE: High-layer Compatibility (len = 4) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Making new call for cr 16976 Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing Q.931 Call Setup Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 4 (cs0, Bearer Capability) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 24 (cs0, Channel Identification) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 30 (cs0, Progress Indicator) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 108 (cs0, Calling Party Number) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 112 (cs0, Called Party Number) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 125 (cs0, High-layer Compatibility) Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Extension '1004' in context 'from-pstn-deviate-custom' from '7034' does not exist. Rejecting call on channel 1/11, span 1 Nov 2 17:22:28 VERBOSE[11329] logger.c: NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Nov 2 17:22:28 VERBOSE[11329] logger.c: Protocol Discriminator: Q.931 (8) len=9 Nov 2 17:22:28 VERBOSE[11329] logger.c: Call Ref: len= 2 (reference 16976/0x4250) (Terminator) Nov 2 17:22:28 VERBOSE[11329] logger.c: Message type: RELEASE COMPLETE (90) Nov 2 17:22:28 VERBOSE[11329] logger.c: [08 02 81 81] Nov 2 17:22:28 VERBOSE[11329] logger.c: Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Nov 2 17:22:28 VERBOSE[11329] logger.c: Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal
Re: [asterisk-users] Remote IP Phone's
Un-top-posting... [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Connor Spiess I am wondering what people are doing for security when registering IP phone's remotely if you do not have the equipment to do a VPN tunnel at the remote site. The phone I would be working with mainly is the Polycom lineup. On Mon, 2 Nov 2009, Danny Nicholas wrote: You can restrict on IP address, MAC address and port type and that's just what I know. If someone want's through bad enough you're going to have a problem, but you can at least slow down or stop casual hackers. Aren't MAC addresses only available to devices on the local network? Don't you mean port number instead of port type? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote IP Phone's
Actually, both. You can (AFAIK) specify 5060, 1 etc and UDP/TCP, etc. Of course, I have been wrong at least once before :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, November 02, 2009 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote IP Phone's Un-top-posting... [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Connor Spiess I am wondering what people are doing for security when registering IP phone's remotely if you do not have the equipment to do a VPN tunnel at the remote site. The phone I would be working with mainly is the Polycom lineup. On Mon, 2 Nov 2009, Danny Nicholas wrote: You can restrict on IP address, MAC address and port type and that's just what I know. If someone want's through bad enough you're going to have a problem, but you can at least slow down or stop casual hackers. Aren't MAC addresses only available to devices on the local network? Don't you mean port number instead of port type? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Remote Party ID
On Mon, Nov 02, 2009 at 03:18:03PM -, Dan Journo wrote: Does anyone know how to set the remote party id? I guess someone does. If you provide more details you'll have better chances of getting a good answer. Version of Asterisk? Why do you want to set the remote party ID? How have you tried setting it so far? Reading the following may give you some hints: http://www.catb.org/~esr/faqs/smart-questions.html -- 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] Asterisk 1.4 and Fax
On 2 Nov 2009, at 17:22, Dan Journo wrote: Does anyone have an up to date guide for setting up fax 2 email with asterisk? So you can fax them obnoxiously long signatures? ___ -- 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] Remote IP Phone's
Un-top-posting... On Mon, 2 Nov 2009, Danny Nicholas wrote: You can restrict on IP address, MAC address and port type and that's just what I know. If someone want's through bad enough you're going to have a problem, but you can at least slow down or stop casual hackers. Aren't MAC addresses only available to devices on the local network? Don't you mean port number instead of port type? On Mon, 2 Nov 2009, Danny Nicholas wrote: Actually, both. You can (AFAIK) specify 5060, 1 etc and UDP/TCP, etc. Of course, I have been wrong at least once before :) I thought SIP on Asterisk was still limited to UDP. I must have missed the memo... -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
We want to disconnect our PSTN fax line and transfer the number over to our asterisk server. I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 November 2009 17:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 and Fax On 2 Nov 2009, at 17:22, Dan Journo wrote: Does anyone have an up to date guide for setting up fax 2 email with asterisk? So you can fax them obnoxiously long signatures? ___ -- 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] Asterisk 1.4 and Fax
Sorry Steve, Forgot to remove it before sending the email. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 November 2009 17:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 and Fax On 2 Nov 2009, at 17:22, Dan Journo wrote: Does anyone have an up to date guide for setting up fax 2 email with asterisk? So you can fax them obnoxiously long signatures? ___ -- 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] Remote IP Phone's
On Monday 02 November 2009 11:46:49 am Steve Edwards wrote: Un-top-posting... On Mon, 2 Nov 2009, Danny Nicholas wrote: You can restrict on IP address, MAC address and port type and that's just what I know. If someone want's through bad enough you're going to have a problem, but you can at least slow down or stop casual hackers. Aren't MAC addresses only available to devices on the local network? Don't you mean port number instead of port type? On Mon, 2 Nov 2009, Danny Nicholas wrote: Actually, both. You can (AFAIK) specify 5060, 1 etc and UDP/TCP, etc. Of course, I have been wrong at least once before :) I thought SIP on Asterisk was still limited to UDP. I must have missed the memo... No, you probably missed that some people are using 1.6, in which TCP and TLS are now available for SIP. If you aren't using 1.6, then TCP and TLS are not available options for SIP. -- Tilghman ___ -- 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] Execute Macro AFTER connecting to a channel
Is the a way to execute the macro AFTER connecting to the channel: M(x[^arg]) - Execute the Macro for the *called* channel before connecting to the calling channel. doesn't work for me as I need to listen to the macro progress as it is sending DTMF tone and respond from the connected channel. -- 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] Asterisk 1.4 and Fax
Dan Journo wrote: I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Faxing over SIP is never a good idea. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] DAHDI/ZAP overlap dialing
--- On Mon, 11/2/09, Martin asteriskl...@callthem.info wrote: I'd double check that you really have overlapdial=yes for those channels ... it should be declared before channel = keyword in zapata.conf/chan_dahdi.conf I declared overlapdial in zapata.conf: switchtype = euroisdn signalling = pri_cpe overlapdial=yes context=from-alcatel-custom group = 1 callgroup = 1 pickupgroup = 1 immediate=no echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=2.0 txgain=1.0 busydetect=no facilityenable = yes ; pritransfer = ect ; either no, ect, or hangup channel = 1-15,17-31 Will try to change libpri versions or move to another 1.4 * server. Thanks for your time. Vieri ___ -- 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] Cannot make calls
Thank you , I did that sip set debug ip 198.163.0.103 SIP Debugging Enabled for IP: 198.163.0.103 I checked the /var/log/asterisk files and there is no information there. Could you please inform where am I suppose to see the debug information ? tks Jair Ott Rose wrote: you can get debug info a couple of ways from the asterisk CLI. I like this command the best. sip set debug ip xxx.xxx.xx.xxx where xxx.xxx.xxx.xxx is the of the x-lite phone. It will give you a lot of info. I haven't figured out how to redirect output yet. Date: Fri, 30 Oct 2009 13:05:35 -0700 From: cliconn...@cliconnect.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cannot make calls No change on it. Do I have to enter a command ? I have changed the port to 5060 in both clients. Still the same problem. thanks Jair Santos Warren Selby wrote: You're attempting to connect on ports 5061-5062 but are bound to port 5060...? What does your CLI look like during a failed call attempt? Thanks, --Warren Selby On Fri, Oct 30, 2009 at 2:18 PM, Cliconnect cliconn...@cliconnect.com wrote: Thank you, How are you setting up xlite and the ata? Xlite User name : 1000 Domain: IP of the server running Asterisk Register with domain and receive incoming calls: clear Port used in local computer : manually specify range : 5061-5062 ATA SIP server address: IP of the server running Asterisk Outbond Proxy : IP of the server running Asterisk SIP User id : 1001 Accoount ID : 1001 Use DNS SRV : yes User id is phone number : yes SIP registration : no Local sip port : 5062 Which version of Asterisk are you using? Asterisk 1.6.1.6, Copyright (C) 1999 - 2009 Digium, What does the general section of your sip.conf look like? [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes When I sip show peers Name/username Host Dyn Nat ACL Port Status 1000 (Unspecified) D 5060 Unmonitored 1001 (Unspecified) D 5060 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] regards Jair Santos Warren Selby wrote: How are you setting up xlite and the ata? Which version of Asterisk are you using? What does the general section of your sip.conf look like? On Fri, Oct 30, 2009 at 1:01 PM, Cliconnect cliconn...@cliconnect.com wrote: Hi all, I can only get a line signal when I set the phones to not register with domain . All phones are in the same NAT and I cannot make calls. I am getting "Call failed : Proxy Authentication Required" in Xlite and a busy signal when using an ATA. Here is my extensions.conf [internal] exten = 1000,1,Verbose(1|Extension 1000) ;exten = 1000,n,Echo() ;exten = 1000,n,Hangup() exten = 1000,n,Dial(SIP/1000,30) exten = 1000,n,Hangup() exten = 1001,1,Verbose(1|Extension 1001) exten = 1001,n,Dial(SIP/1001,30) exten = 1001,n,Hangup() [phones] include = internal and sip.conf [1000] type=friend context=phones host=dynamic [1001] type=friend context=phones host=dynamic I am not setting a password . Any help will be appreciated. TIA Jair Santos -- ___ -- 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 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.698 / Virus Database: 270.14.39/2469 - Release Date: 10/30/09 00:52:00 ___ -- 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 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.698 / Virus Database: 270.14.39/2469 - Release Date: 10/30/09 00:52:00 New Windows 7:
Re: [asterisk-users] Asterisk 1.4 and Fax
I've heard mixed reports. Some say they've had no problems, some say that faxes fail most of the time. I want to try it out and see how it goes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: 02 November 2009 18:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 and Fax Dan Journo wrote: I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Faxing over SIP is never a good idea. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Asterisk 1.4 and Fax
I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. Lee. Dan Journo wrote: I've heard mixed reports. Some say they've had no problems, some say that faxes fail most of the time. I want to try it out and see how it goes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: 02 November 2009 18:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 and Fax Dan Journo wrote: I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Faxing over SIP is never a good idea. Doug ___ -- 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 1.4 and Fax
2009/11/2 Doug Lytle supp...@drdos.info Dan Journo wrote: I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Faxing over SIP is never a good idea. And why would that be? I think that faxing over SIP using T.38 is a fantastic idea. Chris ___ -- 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] Unexpected control subclass '-1'
I have been getting the following message every time I make a call for the past few months: [Nov 2 13:08:18] WARNING[9859]: file.c:1273 waitstream_core: Unexpected control subclass '-1' Everything seems to be working so I do not know if this is important. I am using Asterisk 1.4.26.1 (upgrading today to .2) with Asterisk Addons 1.4.9, Zaptel 1.4.12.1, Libpri 1.4.10 (upgrading to 1.4.10.1 today). What does this message mean? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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 1.4 and Fax
Lee Howard wrote: I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) -- 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] Asterisk 1.4 and Fax
Kevin P. Fleming wrote: Lee Howard wrote: I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) Does Asterisk 1.4 support T.38? Thanks, Lee. ___ -- 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 1.4 and Fax
Lee Howard wrote: FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) Does Asterisk 1.4 support T.38? Only for passthrough between SIP channels; Asterisk 1.6.0 and later also support T.38 termination and origination. -- 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] Asterisk 1.4 and Fax
Christian Victor wrote: 2009/11/2 Doug Lytle supp...@drdos.info mailto:supp...@drdos.info Faxing over SIP is never a good idea. And why would that be? I think that faxing over SIP using T.38 is a fantastic idea. As far as I know, T.38 isn't supported under 1.4 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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 1.4 and Fax
Kevin P. Fleming wrote: Lee Howard wrote: I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) True, although I've yet to find a provider in this country (UK) that supports T.38. He may be better off porting the number to a fax2email service (although ime they are worth play testing first before you put any real work on them, eg. recently I've found one that doesn't support Fine Print or higher res faxes). AFAICT, to get a (real) fax machine using T.38, you either need to buy one that already supports it (never seen one, but I am assured they exist), Buy an ATA that supports it, or move to callweaver. ___ -- 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 1.4 and Fax
How do these fax2email providers run their service? Do they all use physical lines rather than use the internet? Thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon Sent: 02 November 2009 20:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 and Fax Kevin P. Fleming wrote: Lee Howard wrote: I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) True, although I've yet to find a provider in this country (UK) that supports T.38. He may be better off porting the number to a fax2email service (although ime they are worth play testing first before you put any real work on them, eg. recently I've found one that doesn't support Fine Print or higher res faxes). AFAICT, to get a (real) fax machine using T.38, you either need to buy one that already supports it (never seen one, but I am assured they exist), Buy an ATA that supports it, or move to callweaver. ___ -- 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] Asterisk 1.4 and Fax
On Mon, Nov 2, 2009 at 12:22 PM, Dan Journo d...@keshercommunications.comwrote: Hi, Does anyone have an up to date guide for setting up fax 2 email with asterisk? You can buy this shrink-wrapped from Cisco if you're willing to pay what they're asking. There are probably other vendors who sell that too. If you insist on doing this yourself, and using asterisk, start by moving to 1.6. The fax support is night and day better in 1.6 than 1.4, using native asterisk app_fax (which depends on SpanDSP from Lee Howard). If you want to go SIP as part of the deployment, I recommend either: 1) terminate PSTN at your premise, and only use SIP internally inside your PSTN gateway 2) if you're going to go with a SIP provider, tunnel them on a dedicated circuit so you're not fighting bandwidth limit in addition to the various problems you'll inevitably face with their implementation of fax over voip. Once you price #2 you'll probably discover that #1 is cheaper, and I've already said it's more likely to be reliable when you can control as much of the voip as possible. ___ -- 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 1.4 and Fax
David Backeberg wrote: On Mon, Nov 2, 2009 at 12:22 PM, Dan Journo d...@keshercommunications.com mailto:d...@keshercommunications.com wrote:asterisk app_fax (which depends on SpanDSP from Lee Howard). SpanDSP was written by Steve Underwood. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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 1.4 and Fax
On Mon, Nov 2, 2009 at 3:38 PM, Dan Journo d...@keshercommunications.com wrote: How do these fax2email providers run their service? Do they all use physical lines rather than use the internet? If you read far enough back in the archives, you'll find somebody who claimed they used asterisk-1.4 (I think hylafax) and voip But that they did so in a colo, one-hop and almost no RTT away from their provider. Again, at which point, you're not saving money compared to an analogue fax over PSTN unless you have a really large volume, and even then you can often get better bulk pricing for PSTN. You know your usage and you know your budget. If you don't have time to fight broken faxes, learn asterisk-1.6, and provision a voip provider, just stick with analogue fax over PSTN. My business situation: channelized DS3, that's 28x 23 voice channels - Cisco voice routers - SIP - asterisk-1.6 app_fax() Working very well for us, but I don't know whether your budget or usage is going to justify something like that. As for what a commercial service uses, they use whatever was the cheapest wherever they host their services. Real modem pools, or real brooktrout modem boards are common. That would have been a better idea for my situation if I wasn't sharing the circuits with other voice services. ___ -- 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] DTMF Timing and Fujitsu F9600 Switch
We're in the process of replacing an ancient Centigram-based, but Fujitsu-labelled, voicemail system with an Asterisk solution. The system will interface with a Fujitsu F9600 switch and use the SMDI module in Asterisk 1.6.1.x to communicate the calling information needed to make the interplay between the two systems work smoothly. This portion of the project is configured and working fine, as is the programming in extensions.conf. The SMDI work done by Russell at Digium came out really well (thanks). We are having one problem, though. That is, the DTMF interactions are not working consistently from the Fujitsu to Asterisk (particularly the sending of extension number and passcode). Sometimes the system will get just a portion of the extension number or the password and hangup the call before the entering is finished. Sometimes it will work fine, especially if one dials slowly. I have to figure that these problems are resulting from a discordance between the DTMF timing settings on the Fujitsu and those on the Asterisk box. However, I am unsure of how to figure out what settings to use. I can certainly view the settings on the Fujitsu, but do not know of how to tune the DAHDI settings on the Asterisk side. The options looked limited. Or, conversely, what settings does Asterisk expect to see? If I knew that, I could tune on the Fujitsu side of the equation. BTW, the telephony hardware we are using is a Digium Wildcard AEX800 Board 1. Thanks for any help any of you can provide. - Will ___ -- 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] Tutorial for SIP user
Ciao, I installed Xlite on Windows Vista, the IP connection (ping) is working, shall I check something else ? Thanks in advance. 2009/11/1 Farooq Hussain farooqhussain...@gmail.com Dear Giancarlo, On which OS your are installing XLITE. If you are trying to connect XLITE using Winodws XP please make a entry in your firewall. I think that would solve your problem On Sun, Nov 1, 2009 at 10:27 AM, giancarlo lombardo gianclomba...@gmail.com wrote: Dear all, I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have trouble, I see on XLITE console: Registration Error: 503 - Service unavailable. Someone have a tutorial or a step by step description how to do that ? Thanks in advance -- Giancarlo Lombardo ___ -- 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 -- Thanks Farooq Hussain ___ -- 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 -- Giancarlo Lombardo ___ -- 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] Anyone seen this before
Testing a new gateway and have a Rhino Channel Bank... Sending a test fax and everything works fine (Receive the fax fine) But I notice this in the log Google search didn't return much of anything... DAHDI hook failed returned -1 (trying 1): Device or resource busy ___ -- 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] Anyone seen this before
On Mon, Nov 2, 2009 at 4:26 PM, Robert Grignon rgrig...@fleetone.com wrote: Testing a new gateway and have a Rhino Channel Bank... Sending a test fax and everything works fine (Receive the fax fine) But I notice this in the log Google search didn't return much of anything... DAHDI hook failed returned -1 (trying 1): Device or resource busy I've never used the channel bank, but different types of phones do different things to signal a hangup, including perhaps temporarily reversing polarity. Whichever way your card is configured to detect hangups, try switching the line signaling and see if that fixes your problems. ___ -- 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 1.4 and Fax
Dan Journo wrote: How do these fax2email providers run their service? I've not the faintest Idea, the provider I use afaict outsource it. Do they all use physical lines rather than use the internet? Thanks Dan ___ -- 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 as Outbound Proxy ?
Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone - via TLS/SRTP - Asterisk as outbound proxy - via UDP/RTP - VoIP-Provider So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER) to the VoIP-Provider and do SIP TLS- SIP UDP and SRTP - RTP translation (via *1.6.2 and the SRTP patch) Kristijan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as Outbound Proxy ?
Kristijan Vrban wrote: Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone - via TLS/SRTP - Asterisk as outbound proxy - via UDP/RTP - VoIP-Provider So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER) to the VoIP-Provider and do SIP TLS- SIP UDP and SRTP - RTP translation (via *1.6.2 and the SRTP patch) It is highly unlikely that you'll be able to get Asterisk configured in a transparent enough fashion to appear as a proxy in this scenario. You'd be far better off to use an actual proxy, if that's the functionality you need. -- 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] Asterisk 1.4 and Fax
2009/11/2 Doug Lytle supp...@drdos.info Christian Victor wrote: 2009/11/2 Doug Lytle supp...@drdos.info mailto:supp...@drdos.info Faxing over SIP is never a good idea. And why would that be? I think that faxing over SIP using T.38 is a fantastic idea. As far as I know, T.38 isn't supported under 1.4 That would be Faxing using Asterisk 1.4 is never a good idea. Sorry for being such a bean counter. ;-) To stay on-topic: Terminating fax over PSTN works quite well in 1.4 but the original poster should be warned of trying to terminate fax over a SIP trunk. Using SIP/G.711 to connect the fax machine to Asterisk over LAN works quite well in my experience but others had worse results. Chris ___ -- 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 1.4 and Fax
On 11/03/2009 04:25 AM, Thomas Kenyon wrote: Kevin P. Fleming wrote: Lee Howard wrote: I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) True, although I've yet to find a provider in this country (UK) that supports T.38. He may be better off porting the number to a fax2email service (although ime they are worth play testing first before you put any real work on them, eg. recently I've found one that doesn't support Fine Print or higher res faxes). AFAICT, to get a (real) fax machine using T.38, you either need to buy one that already supports it (never seen one, but I am assured they exist), Buy an ATA that supports it, or move to callweaver. T.38 FAX machines do exist, although they are rare. A number of high end office machines support T.38, or have a T.38 option. There are small FAX machines from Sagem which support T.38. 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] Real replacement for AgentCallBackLogin() on Asterisk 1.6
My mental plan orginilly was: 1.- Creating a macro that acceps ARGs like. a.- agent b.- queue/s In the macro we could have the voice respose for the loging. I am using on 1.4 the following procedure. * the agents call to 21Agentid to loging, and it is promt just for the passwd * the agents call to 22Agentid to logoff using the same philosofy we could implement some easy marco that only ask for the password and: 1.- sets the astdb 2.- sets the globals AGENTBYCALLERID_X= 3.- adds the agent to the queues. Let me work deeper on this idea and see what comes up. ML 2009/11/2 Lenz Emilitri lenz.lo...@gmail.com We were thinking about doing something similar as well. A lot of people are asking for this. If there is anybody else interested, we could share the load I was thinking about creating a context like @agents, so that when you do the log-on you basically add Local/1...@agents as a member of the queue. When you ring it, it basically looks up for the actual device in AstDB and dials it like: Queue - (member) Local/1...@agents - (astdb) SIP/234 I think that we should be able to forward channel state as well (using hints? I've never done it) so that app_queue does not try dialling agents that are busy. I was thinking about storing queue-agent associations into config strings, and/or AstDB, and/or http over curl. And yes, ideally it should work fine on 1.4's as well Things that should be working from version one: - logging compatible with older asterisk's - authentication using Voicemail -.plug and play on most systems - channel states - pause/unpause with pause codes - ...you tell me Anybody interested? l. 2009/10/30 Mariano Lecuona mlecu...@gmail.com Hi all, I would like to know if there is any application replacement for the AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from what I've read that the call back login agent can be done using a smart dialplan as showed on the docs. But I cannot thinks a flexible dialplan for a dinamic reassignation of agents to different queues every day. Thanks in advance. Mariano ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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] turn the ring tone OFF during dialing
Is the a way to turn the ring tone OFF during dialing? When I'm in a macro mode I have to listen to ring the tone for 20sec before macro finish and I get connected. -- 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] turn the ring tone OFF during dialing
On 11/02/09 19:56, Joseph wrote: Is the a way to turn the ring tone OFF during dialing? When I'm in a macro mode I have to listen to ring the tone for 20sec before macro finish and I get connected. I've found a better solution, setting musing on hold before calling party answers: m: Provide Music on Hold to the calling party until the called channel answers. but it doesn't seems to work, there in musing during dialing: [goto-dialout] exten = 51,1,SetMusicOnHold(default) exten = 51,n,Dial(SIP/18775898...@pstn-5665,20,m(default)M(atb)) Music on Hold is working OK, so why this one isn't working? -- 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] Asterisk as Outbound Proxy ?
Proxy is not the correct term to describe this scenario, but yes, it is possible in principle. Kristijan Vrban wrote: Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone - via TLS/SRTP - Asterisk as outbound proxy - via UDP/RTP - VoIP-Provider So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER) to the VoIP-Provider and do SIP TLS- SIP UDP and SRTP - RTP translation (via *1.6.2 and the SRTP patch) Kristijan ___ -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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] Core Dump - Asterisk 1.4.24 - Elastix
Hi, Yesterday I've got a core dump from Asterisk, other times I was able to discover what this core dump was related with through gdb Ouput info,, but this time.. I'm really lost. Could some one please help me GDB output is at http://pastebin.com/m603e6a74 Any help would be appreciated. Best Regards, Fernando ___ -- 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] Exchange 2007 UM issues with Asterisk 1.6
Hi Guys, Does anyone know how to make the custom build of Asterisk 1.6.1 work with Exchange 2007 UM? It always times out with system unavailable and won't go through. I have TCP enabled and all the trunk and outgoing settings configured. When I use TrixBox CE 2.8.0.2, it works. But custom install, won't work, everything else works except for Exchange 2007 UM. Any help would be appreciated. Thanks, KJ ___ -- 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] Core Dump - Asterisk 1.4.24 - Elastix
On Tue, Nov 03, 2009 at 01:18:14AM -0300, Fernando Berretta wrote: Hi, Yesterday I've got a core dump from Asterisk, other times I was able to discover what this core dump was related with through gdb Ouput info,, but this time.. I'm really lost. Could some one please help me GDB output is at http://pastebin.com/m603e6a74 When Asterisk is installed from a binary package, you will normally get binaries without debug symbols to save space. You sypically don't need the debug information. However for the cases you do need them, the debug symbols are available in a separate package (with rpm: the *-debuginfo packages , with debs: -dbg packages or several similar things). Debug symbols are not needed at core dumping time. They are only needed when you try to get something useful from a core dump (e.g. with gdb). So you can install them after dumping the core. However they must be of exacatly the same version of the Asterisk package installed on your system (that dumped the core in the first place). So basically: just install the package asterisk-debuginfo and try aagain. There's also a similar package for glibc for the libc stuff, though this typically is less useful. -- 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
