Re: [Asterisk-Users] IAX clients and the flash button
Hi, - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 4:59 AM Subject: [Asterisk-Users] IAX clients and the flash button Hi guys As usual I am playing around with IAX soft clients. I was wondering with the various IAX clients, IAX client, DIAX, etc how's one park calls, transfer calls, etc since there is no flash key? With DIAX you can transfer using '#' key. Then transfer the call to 701 You will be able to park by transfering to 701 (or whatever number you have defined). It works. You do not need Flash key for a soft phone. Is there something I must do in the iax.conf or is it something I must do with the individual clients? Nothing. Also, is it very difficult to use musiconhold with the IAX software clients? MOH works too. Have you tried??? Thanks a lot for any ideas, suggestions, feedback. AJ Please give me your feedback with DIAX regarding those issues. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P - module does not gat loaded
Hi I installed a x100P card today. Once it is configured * no longer starting. It gives me the following error. == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1074432736]: File chan_zap.c, Line 629 (zt_open): Unable to specify chan nel 1: No such device or address ERROR[1074432736]: File chan_zap.c, Line 4973 (mkintf): Unable to open channel 1 : No such device or address here = 0, tmp-channel = 0, channel = 1 ERROR[1074432736]: File chan_zap.c, Line 6755 (load_module): Unable to register channel '1' WARNING[1074432736]: File loader.c, Line 305 (ast_load_resource): chan_zap.so: l oad_module failed, returning -1 WARNING[1074432736]: File loader.c, Line 400 (load_modules): Loading module chan _zap.so failed! [EMAIL PROTECTED] asterisk]# here is my zapata.conf [EMAIL PROTECTED] asterisk]# cat zapata.conf [channels] language=en context=analog-in signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 1 ouput of lsmod [EMAIL PROTECTED] asterisk]# lsmod Module Size Used byNot tainted iptable_filter 2412 0 (autoclean) (unused) ip_tables 15096 1 [iptable_filter] wcfxo 9056 0 (unused) zaptel181504 0 [wcfxo] ppp_generic2 0 [zaptel] slhc6740 0 [ppp_generic] nls_iso8859-1 3516 1 (autoclean) udf98400 0 (autoclean) ide-cd 35708 1 (autoclean) cdrom 33728 0 (autoclean) [ide-cd] parport_pc 19076 1 (autoclean) lp 8996 0 (autoclean) parport37056 1 (autoclean) [parport_pc lp] autofs 13268 0 (autoclean) (unused) 3c59x 30704 1 cs4232 5444 0 ad1848 28556 0 [cs4232] uart401 8388 0 [cs4232] sound 74228 0 [cs4232 ad1848 uart401] soundcore 6404 4 [sound] keybdev 2944 0 (unused) mousedev5492 1 hid22148 0 (unused) input 5856 0 [keybdev mousedev hid] usb-uhci 26348 0 (unused) usbcore78784 1 [hid usb-uhci] ext3 70784 2 jbd51892 2 [ext3] and [EMAIL PROTECTED] asterisk]# ztcfg -vv Zaptel Configuration == Channel map: 0 channels configured. Anyone can shed some light here. Cheers Sathya ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAX users
Hi all, For the DIAX users who use the CallMe feature in the Help menu, I kindly ask you that if you call, to leave a message too. I have a lot of calls using this feature and there is just a click. The user hangup without leaving any message. Thank you for your understanding, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P - module does not gat loaded
quote who=Sathya Weerasooriya I installed a x100P card today. Once it is configured * no longer starting. [snip] [EMAIL PROTECTED] asterisk]# ztcfg -vv Zaptel Configuration == Channel map: 0 channels configured. Have you configured /etc/zaptel.conf? -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P - module does not gat loaded
Did you setup your zaptel.conf? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Sathya Weerasooriya Sent: Wednesday, November 05, 2003 2:14 AM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Users] X100P - module does not gat loaded Hi I installed a x100P card today. Once it is configured * no longer starting. It gives me the following error. == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1074432736]: File chan_zap.c, Line 629 (zt_open): Unable to specify chan nel 1: No such device or address ERROR[1074432736]: File chan_zap.c, Line 4973 (mkintf): Unable to open channel 1 : No such device or address here = 0, tmp-channel = 0, channel = 1 ERROR[1074432736]: File chan_zap.c, Line 6755 (load_module): Unable to register channel '1' WARNING[1074432736]: File loader.c, Line 305 (ast_load_resource): chan_zap.so: l oad_module failed, returning -1 WARNING[1074432736]: File loader.c, Line 400 (load_modules): Loading module chan _zap.so failed! [EMAIL PROTECTED] asterisk]# here is my zapata.conf [EMAIL PROTECTED] asterisk]# cat zapata.conf [channels] language=en context=analog-in signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 1 ouput of lsmod [EMAIL PROTECTED] asterisk]# lsmod Module Size Used byNot tainted iptable_filter 2412 0 (autoclean) (unused) ip_tables 15096 1 [iptable_filter] wcfxo 9056 0 (unused) zaptel181504 0 [wcfxo] ppp_generic2 0 [zaptel] slhc6740 0 [ppp_generic] nls_iso8859-1 3516 1 (autoclean) udf98400 0 (autoclean) ide-cd 35708 1 (autoclean) cdrom 33728 0 (autoclean) [ide-cd] parport_pc 19076 1 (autoclean) lp 8996 0 (autoclean) parport37056 1 (autoclean) [parport_pc lp] autofs 13268 0 (autoclean) (unused) 3c59x 30704 1 cs4232 5444 0 ad1848 28556 0 [cs4232] uart401 8388 0 [cs4232] sound 74228 0 [cs4232 ad1848 uart401] soundcore 6404 4 [sound] keybdev 2944 0 (unused) mousedev5492 1 hid22148 0 (unused) input 5856 0 [keybdev mousedev hid] usb-uhci 26348 0 (unused) usbcore78784 1 [hid usb-uhci] ext3 70784 2 jbd51892 2 [ext3] and [EMAIL PROTECTED] asterisk]# ztcfg -vv Zaptel Configuration == Channel map: 0 channels configured. Anyone can shed some light here. Cheers Sathya ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outband DTMF on i4l modem
Hello, I am setting up 2 ISDN 4 linux cards and have had great success now that I have got over the initial problems with : and / characters. The only problem I am experiencing now is the sending of DTMF tones over the line to a remote IVR system. If I dial SIP (Cisco 7905 and 7940) to a number over the line, no DTMF tones are heard. I dialed my own home phone and tried it, no matter which button i pressed, no tones came out. However when I dial voicemail, the buttons work fine. Is this a problem with asterisk not putting the tones onto i4l? Thanks, Matthew Enger [EMAIL PROTECTED] Xintegration ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Client Dev - Newbie questions
Hi, Is it possible to develop a client (IAX/SIP/H323) and work from inside a browser (IE/NS/MG) for CRM reasons ? Any work around ? Suggestions ? Is there a manager console over a browser ? If not, is there an intention to develop one ? Also, is there a TCP/IP server to control * in order to supportour up layer predictive dialer ? MarinBlu Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard
Re: [Asterisk-Users] Call Transfert with SwissVoice IP10S in MGCP mode
I have the MGCP Firmware and call transfer doesn't work in my configuration. Daniel Marian Danisek a écrit: Daniel ANDRE wrote: Hello, Now that I have a nearly working configuration for my IP10S with * I wonder if anyone has done call transfert with this Phone. In the IP10S documentation they talk about the 'service key' wich is the key with the white dot on it. With this Key, it should be possible to have a menu with call transfert entries. This menu should (accordingly to the documentation) depend on the call manager. In my case, I have the message 'No available service' instead. What's wrong? Daniel call transfer in ip10s is possible only with mgcp formware... my phones works with h323... so no way... read notes from support : For the moment, call transfer is not yet fully integrated, so not proposed through the man machine interface. Call transfer will be ma naged through H.450. -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfert with SwissVoice IP10S in MGCP mode
Daniel ANDRE wrote: I have the MGCP Firmware and call transfer doesn't work in my configuration. show mgcp.conf Daniel Marian Danisek a écrit: Daniel ANDRE wrote: Hello, Now that I have a nearly working configuration for my IP10S with * I wonder if anyone has done call transfert with this Phone. In the IP10S documentation they talk about the 'service key' wich is the key with the white dot on it. With this Key, it should be possible to have a menu with call transfert entries. This menu should (accordingly to the documentation) depend on the call manager. In my case, I have the message 'No available service' instead. What's wrong? Daniel call transfer in ip10s is possible only with mgcp formware... my phones works with h323... so no way... read notes from support : For the moment, call transfer is not yet fully integrated, so not proposed through the man machine interface. Call transfer will be ma naged through H.450. -- - Best Regards, Pavel Litvinenko. ICQ: 16224754 Ph: (8632) 923962, 923640 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hello Dan, Its an excellent start. Please don't get swayed away by some stupid remarks. I am really impressed by your work and I hope to see a lot more releases from you. In spirit of improving the code, here are some of the issues that I faced while trying it out: 1. Once I dial the number, the directory disappears and never shows up. 2. I see a message at the bottom that says dialiing and the message goes away after a second or so. But nothing happens and the Dial/Hangup Delete button goes gray on me. 3. It will be nice to see the status at the bottom at all times and have a hang-up button (even though the user may not be in call). This is just what we are used to doing on regular phone line. If something is wrong, click the falsh button a couple of time and hope that things would improve. That's it from me so far. Overall, great work and thanks a lot for all your efforts. Please keep it up. Ricky -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Masakazu Nakano Sent: Sunday, November 02, 2003 7:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Hi Dan. thanks for good application! and I wish 'no with installer' package about that. because I think use with USB-memory device in any places (ie.net-cafe.) is that need registry setting or not? On Sun, 2 Nov 2003 22:21:09 +0200 Dan [EMAIL PROTECTED] wrote: Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *, Fritz!PCI and strange behavior
I'm testing * (CVS-09/16/03-02:07:49 with zaprtc 0.0.1) with Fritz!PCI (chan_capi 0.3.0), and have a couple of funny things - I wonder if anyone else has seen them: Hmm, I'm running plain vanilla * v0.5 and have no problems with that particular card, same version of chan_capi. Did you compile fcpci driver yourself? I'm on RH9. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
Some more remarks: 1: Message: Unknown event: 6 for call 1 2. Message: No free call appearences ? 3. Again, two buttons grayed out? ricky -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, November 05, 2003 12:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform) Hello Dan, Its an excellent start. Please don't get swayed away by some stupid remarks. I am really impressed by your work and I hope to see a lot more releases from you. In spirit of improving the code, here are some of the issues that I faced while trying it out: 1. Once I dial the number, the directory disappears and never shows up. 2. I see a message at the bottom that says dialiing and the message goes away after a second or so. But nothing happens and the Dial/Hangup Delete button goes gray on me. 3. It will be nice to see the status at the bottom at all times and have a hang-up button (even though the user may not be in call). This is just what we are used to doing on regular phone line. If something is wrong, click the falsh button a couple of time and hope that things would improve. That's it from me so far. Overall, great work and thanks a lot for all your efforts. Please keep it up. Ricky -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Masakazu Nakano Sent: Sunday, November 02, 2003 7:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Hi Dan. thanks for good application! and I wish 'no with installer' package about that. because I think use with USB-memory device in any places (ie.net-cafe.) is that need registry setting or not? On Sun, 2 Nov 2003 22:21:09 +0200 Dan [EMAIL PROTECTED] wrote: Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Demo Weather Report AGI v2.0
Some of you may know me as ManxPower from #Asterisk at irc.freenode.,net I've posted my demp weather report Asterisk AGI script at http://www.fnords.org/~eric/asterisk/downloads/ Eric, Can you comment on the difference in installation ease for Festival and Cepstral? Regards, Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, Nov 04, 2003 at 07:24:19PM +0100, Olle E. Johansson wrote: Keep feeding the list, I'll steel information to the wiki. http://www.voip-info.org/tiki-index.php?page=Asterisk+setup+medium+office Superb stuff, Olle :) If we can establish a 'standard format' (maybe an HTML form?) for configuration postings, perhaps people will be more likely to submit their data? Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, Nov 04, 2003 at 01:52:46PM -0600, Steven Critchfield wrote: - Has been in nearly fault free operation for more than since 05-2002. Great stuff, Steven! :) Can I enquire what was the cause of the downtime? Was it planned- maintenance, or an actual fault with the Asterisk software / Digium hardware? Roughly how long has the system been down in total since 'going live' ? Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and NAT: try, try again.
In response to the SIP and NAT discussion, I have updated the ticket on the subject that seemed to be getting the most attention: #104. There are enough clueful people here that perhaps someone can come up with a patch that handles NAT in the elegant way that I describe in the bugnotes, as I am but a mere integrator who has limited C skills. In the absence of such a patch being offered, we await William Waites' patch and disclaimer which will at least be more sufficient than the current externip= method. Those with an interest in the discussion of how Asterisk should handle being put behind a NAT should direct their attention to: http://bugs.digium.com/bug_view_page.php?bug_id=104 JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi, - Original Message - From: Asterisk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 10:55 AM Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform) Some more remarks: 1: Message: Unknown event: 6 for call 1 This message is generated by the IAX library. I don't know why (yet) By the way... in the final release this statusbar will be available only in administrative mode, not accessible for a regular user. 2. Message: No free call appearences ? The same. 3. Again, two buttons grayed out? Which? If talk about the two function buttons, then they are grayed out if: - no call was placed durring the current application execution - - no active call and no redial information available There is another situation when they are grayed out? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snatching calls
It works with SIP and with zap channels. What about IAX? like DIAX softphone? I may be misunderstanding something. When you start an Asterisk configuration process, connecting your hardware and building your dialplan, you use Zapata.conf, sip.con and iax.conf to connect the FXSs and FXOs to your Asterisk, as you would connect your normal lines and extensions to a regular PBX. The extensions.conf is used not only to devise the Dial plan, but also to assign extension numbers to all the extensions connected (be it sip, iax, zap, or whatever). Now I get to the groups. Call groups. Pickup groups, whatever groups. Isn't it logical that it would be done in extensions.conf? or in an other single location, and not throughout several files that you have to recheck every time you want to change anything? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Liew Sent: Wednesday, November 05, 2003 6:10 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] snatching calls - Original Message - From: Billy Huddleston [EMAIL PROTECTED] how could you do this with sip and VOIP? From: Steven Critchfield [EMAIL PROTECTED] You want to look into call groups and pickup groups. To pickup the call you use *8#. from /usr/src/asterisk/configs/zapata.conf.sample ; ; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing ; and it is a member of a group which is one of your pickup groups, then ; you can answer it by picking up and dialing *8#. For simple offices, just ; make these both the same ; callgroup=1 pickupgroup=1 As per what Steve says, the same applies for SIP, check /usr/src/asterisk/configs/sip.conf.sample Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alert extensions without answering incoming call?
On Tue, 4 Nov 2003 14:06:43 -0800 John Todd [EMAIL PROTECTED] wrote: Hi, * gurus, I wonder if there is a way to alert the in-house extension(s) in case of an incoming external call without actually answering it, before somebody picks up the phone on one of the extensions? This way the caller wouldn't have to pay for the call until somebody answers. Regards, Christian Lademann The Cisco 79xx series phones in SIP mode have a distinctive ring that can be triggered from within Asterisk using the ALERT_INFO variable. They are the only phones that I am aware of that support Hello, John, thank you for your answer. How would this feature be used from extensions.conf? (Sorry, if this looks like a newbie-question, in fact, it is one :-) Regards, Christian Lademann this SIP standard variable. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and NAT: try, try again.
...and to solve another problem, there's my suggestion on support for outbound SIP proxy. http://bugs.digium.com/bug_view_page.php?bug_id=359 There are corporate networks that use a SIP proxy proxy as an ALG, application layer gateway, for all outbound and inbound SIP traffic in the DMZ. This should work in conjunction with netmask/STUN - if host does not belong to my network send SIP transaction to outbound proxy else send SIP transaction to host done This cleverness may cause problems with inside networks consisting of several networks with different netmasks and complicated routing... I believe outbound proxy should be configured on a host by host basis for sip clients/peers as well as an default outbound proxy to use in other situations. In order to support SIP URL dialling, we have to use a netmask/STUN solution to sort out if the SIP proxy we're trying to reach is ourself, someone on the inside or someone on the outside of our NAT. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.skype.com/
Hi! http://www.skype.com/ seesm to be the latest craze... anyone have any knowledge of their technolgy use etc ?? - closed source - WinXP and 2k only - peer-2-peer, i.e. they route foreign calls through your client (and bandwidth) if that helps the calling parties Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.skype.com/
Philipp von Klitzing wrote: http://www.skype.com/ seesm to be the latest craze... anyone have any knowledge of their technolgy use etc ?? - closed source - WinXP and 2k only - peer-2-peer, i.e. they route foreign calls through your client (and bandwidth) if that helps the calling parties In one of our Swedish daily newspapers, like the national Financial Times, one of the owners said that they're going to sell a commercial version with PSTN connectivity early next year. As I understand it they must not be fully peer-to-peer even if they use your bandwidth, there has to be media servers in their network, handling calls. Or? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.skype.com/
On 05/11/03 10:14, Olle E. Johansson wrote: As I understand it they must not be fully peer-to-peer even if they use your bandwidth, there has to be media servers in their network, handling calls. Or? No, the whole point is that it's completely decentralized. More interesting to end users is that the calls are encrypted and can traverse NAT. The way Skype can bounce between peers effectively enables it to provide a few different routes for the traffic, from which it picks the least latency one. Add a nice UI, and it's not surprising that it's gathering speed rapidly. -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.skype.com/
Philipp von Klitzing wrote: http://www.skype.com/ seesm to be the latest craze... anyone have any knowledge of their technolgy use etc ?? - closed source - WinXP and 2k only - peer-2-peer, i.e. they route foreign calls through your client (and bandwidth) if that helps the calling parties In one of our Swedish daily newspapers, like the national Financial Times, one of the owners said that they're going to sell a commercial version with PSTN connectivity early next year. As I understand it they must not be fully peer-to-peer even if they use your bandwidth, there has to be media servers in their network, handling calls. Or? /O My limited understanding: If you have a public IP address (non-NAT) then you will see more traffic going through your session than most, since NAT'ed hosts need a relay on the outside of their NATs. Skype uses the Global IP Sound codecs, which are tremendously efficient. Voice quality is reportedly excellent, even under the extreme examples of multi-application use dialup connections. Skype encrypts all sessions at the management and media layers, which is a feature that I _love_ and wish Asterisk would develop more robustly. Skype is indeed proprietary, and is a for-profit company, so don't expect a chan_skype to happen soon unless they decide that they want to play nice with others (doubtful.) Skype will certainly be introducing PSTN connectivity, but I am very interested in what their numbering plan will look like for inbound calls, if such a plan is contemplated at all. These guys have to make money, so look for any new features costing $$$ - don't get too hooked yet (Anyone remember the problems .mp3 and .gif formats? Hell?) Skype has the ease of use and features to which we, as the rest of the VoIP community, should aspire. Extremely easy setup, excellent call quality, robust and distributed routing, and secure transmissions. They are certainly lacking many of the features that makes something good, such as compliance with standards, but as a private company they can ignore those issues because they're not doing this for the betterment of anyone but themselves. If we can implement Skype-like features in our software but still develop in the open source, standards-compliant world, then that is a noble goal. Skype will certainly lead the way in showing us what features the customers want, and their system will push us towards making real VoIP networks of a much larger and robust (P2P) scale, but ultimately I think they'll fail due to their closed source methods. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.skype.com/
On Wed, 05 Nov 2003 10:36:01 +, Alastair Maw wrote: On 05/11/03 10:14, Olle E. Johansson wrote: As I understand it they must not be fully peer-to-peer even if they use your bandwidth, there has to be media servers in their network, handling calls. Or? No, the whole point is that it's completely decentralized. More interesting to end users is that the calls are encrypted and can traverse NAT. The way Skype can bounce between peers effectively enables it to provide a few different routes for the traffic, from which it picks the least latency one. Add a nice UI, and it's not surprising that it's gathering speed rapidly. what the question is how without some for of centralisation can they have BOTH ends behind NAT ?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.skype.com/
Alastair Maw wrote: On 05/11/03 10:14, Olle E. Johansson wrote: As I understand it they must not be fully peer-to-peer even if they use your bandwidth, there has to be media servers in their network, handling calls. Or? No, the whole point is that it's completely decentralized. More interesting to end users is that the calls are encrypted and can traverse NAT. The way Skype can bounce between peers effectively enables it to provide a few different routes for the traffic, from which it picks the least latency one. Add a nice UI, and it's not surprising that it's gathering speed rapidly. So all peers exchange traffic constantly over UDP to keep NAT bindings open? And a central server to set it all up... Hmmm. Interesting. If I have a network connection with low latency and use SKype, the risk is that my network bogs down with the automatic routing of calls to my connection... Maybe we should develop IAX3 with automatic p2p routing/latency handling? :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: *, Fritz!PCI and strange behavior
Peter Zeltins [EMAIL PROTECTED] said: Hmm, I'm running plain vanilla * v0.5 and have no problems with that particular card, same version of chan_capi. Did you compile fcpci driver yourself? I'm on RH9. Yes, I compiled it myself. I'm running on Debian unstable, kernel 2.4.21 (homebuild) -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP broken for budgtone.
I just downloaded the newest version from CVS([EMAIL PROTECTED]) and I am getting an error whenever
I call the asterisk box. I cannot here any audio on the budgtone. This works
fine with my pingtel phone and my sip 7960. Also if I call my Skinny 7960 it
rings but I get that same error when I pick up. When the skinny phone calls the
Budgtone it works fine. I have 2 budgtone phones and it does this on both of
them. This worked fine before I installed the newest version of
asterisk.
-- Executing
Playback("SIP/budgtone-7ee9", "carried-away-by-monkeys") in new
stack -- Playing 'carried-away-by-monkeys' (language
'en') -- Executing Playback("SIP/budgtone-7ee9",
"lots-o-monkeys") in new stack -- Playing 'lots-o-monkeys'
(language 'en')WARNING[40966]: File chan_sip.c, Line 456 (retrans_pkt):
Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1735 (Response)
With sip debug
Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
192.168.1.223 From: "William Carlson"
sip:[EMAIL PROTECTED];tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To:
sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 62159 INVITE User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp Content-Length: 263 v=0 o=budgtone 0 0 IN
IP4 192.168.1.223 s=- c=IN IP4 192.168.1.223 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18
2 15 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000
12 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.223 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format ULAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 285/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Reliably Transmitting (no NAT):SIP/2.0 407 Proxy Authentication
Required Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson"
sip:[EMAIL PROTECTED];tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To:
sip:[EMAIL PROTECTED];tag=as67b6f854 Call-ID: [EMAIL PROTECTED]
CSeq: 62159 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk",
nonce="6c3e5732" Content-Length: 0 to 192.168.1.223:5060
Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
192.168.1.223 From: "William Carlson"
sip:[EMAIL PROTECTED];tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To:
sip:[EMAIL PROTECTED];tag=as67b6f854 Contact:
sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED]
CSeq: 62159 ACK User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow:
INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Length: 0
11 headers, 0 lines
Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
192.168.1.223 From: "William Carlson"
sip:[EMAIL PROTECTED];tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To:
sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]
Proxy-Authorization: DIGEST username="budgtone", realm="asterisk",
algorithm=MD5, uri="sip:[EMAIL PROTECTED]", nonce="6c3e5732",
response="4e90c985822b15d83f297e8c4fe80372" Call-ID: [EMAIL PROTECTED]
CSeq: 62160 INVITE User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp Content-Length: 263 v=0 o=budgtone 0 0 IN
IP4 192.168.1.223 s=- c=IN IP4 192.168.1.223 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18
2 15 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000
13 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.223 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format ULAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 285/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for 9998 in default
list_route: hop: sip:[EMAIL PROTECTED]
Transmitting (no NAT):SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.223
From: "William Carlson"
sip:[EMAIL PROTECTED];tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To:
sip:[EMAIL PROTECTED];tag=as5481a27e Call-ID: [EMAIL PROTECTED]
CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0
to 192.168.1.223:5060
[Asterisk-Users] g.729 codec registration
Hi all, i have purchased the g.729 codec from digium. The registration was successful. (with the old binary) But there're a few questions: - should not the codec listed in the codec list when i enter show codecs ? - the codec is named with g729b but if i enter show codecs there is a codec g729a listed also the g729b is not installed. what is the difference between g729a built in * and the puchased g729b codec? Thanks for help. Regards, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfert with SwissVoice IP10S in MGCP mode
Daniel ANDRE wrote: I have the MGCP Firmware and call transfer doesn't work in my configuration. this is my mgcp.conf with working call transfer: [general] port = 2427 bindaddr = 192.168.1.253 [192.168.1.92] threewaycalling=yes transfer=yes callwaiting=yes callwaitingcallerid=yes host=192.168.1.92 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = John 92 line = aaln/1 [192.168.1.91] threewaycalling=yes transfer=yes callwaiting=no callwaitingcallerid=no host=192.168.1.91 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = Mary 91 line = aaln/1 -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] First AGI help..
I am trying to write my first AGI script..
I cant seem to get it to work.. I am trying PHP in preference (I know
this is frowned upon) but I can't get it to work with perl either.. I
guess I just don't understand it correctly..
All I am trying to do is get the script to make a call using Dial so
that I can get an idea of how it works.. My experience is really only in
PHP based websites so this is a whole new ball game for me..
Thanks for any help.. and any samples (even offlist) would be greatly
appreciated..
I have setup in extensions.conf like this for testing..
exten = 7000,1,agi(test1.php)
exten = 7001,1,agi(test2.pl)
test1.php script..
?php
// From Kapjod's sample..
ob_implicit_flush(true);
set_time_limit(0);
$err = fopen(php://stderr,w);
$in = fopen(php://stdin,r);
$out = fopen(php://stdout,w);
//This works..
fputs($out, Verbose \Calling phone\n);
// This doesn't
fputs($out, exec(Dial(sip/2012)\n);
fclose($in);
fclose($out);
fclose($err);
?
test2.pl script..
#!/usr/bin/perl
# taken from a sample file..
$|=1;
while(STDIN) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
#This works..
print STDOUT AGI Environment Dump:\n;
foreach $i (sort keys %AGI) {
print STDOUT Verbose \-- $i = $AGI{$i}\\n;
}
# This does not start a call..
print STDOUT exec(Dial(sip/2012)\n)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [REPOST] [Asterisk-Users] ZapRAS docs needed...
On Wed, 2003-11-05 at 04:55, Roy Sigurd Karlsbakk wrote: Is there any information available about ZapRAS other than the fscking source? just pointing out for those interested in reading that I have written up my recent experience over on the -dev list where I read this first. Ollie, you want to pick it up and find a suitable place for it in the wiki? On Thu, 2003-10-30 at 12:52, Roy Sigurd Karlsbakk wrote: hi all Where can I find documentation about how to setup ZapRAS? What I want to do (optimally) is to allow for automatic dial-up to external sites, each having an ISDN router. Today we use a small ISDN router for this, but it'd be a lot better, IMHO, to have asterisk do this (functioning as a ISDN router), as we may cancel our BRIs then. Is this possible? And if so, how can I do it? I can't find any docs about ZapRAS at all! roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] First AGI help..
Hi ?php // From Kapjod's sample.. ob_implicit_flush(true); set_time_limit(0); $err = fopen(php://stderr,w); $in = fopen(php://stdin,r); $out = fopen(php://stdout,w); //This works.. fputs($out, Verbose \Calling phone\n); // This doesn't fputs($out, exec(Dial(sip/2012)\n); fclose($in); fclose($out); fclose($err); ? You'll find its to do with your syntax - show agi exec produces Usage: EXEC application options Executes application with given options. Returns whatever the application returns, or -2 on failure to find application ie use spaces not (. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Wed, 2003-11-05 at 03:11, Gavin Hamill wrote: On Tue, Nov 04, 2003 at 01:52:46PM -0600, Steven Critchfield wrote: - Has been in nearly fault free operation for more than since 05-2002. Great stuff, Steven! :) Can I enquire what was the cause of the downtime? Was it planned- maintenance, or an actual fault with the Asterisk software / Digium hardware? One failure was a kernel lockup when compiling a new module to start testing ZapRas. I have made a couple of stupid mistakes that shut asterisk down. Outside of the above comments, my gateway machine has worked flawlessly, but this is also why it isn't given much to do. It is too important to have something make it fail. On my pbx machine in the office though, we occasionally have small failures. Most recently we had a segfault show up in a zapata handle event function, but couldn't track it down well enough to report upon it. This machine is specifically set up to be more of a test bed machine. We have only 4 people currently in our office, and 2 of us use the phones mainly for testing of our software. Roughly how long has the system been down in total since 'going live' ? I'd say we haven't had more than 10 minutes downtime on our gateway machine, and thats mostly due to the kernel lockup that caused me to have to call my colo facility to do a hands on reset of the machine. We may have about that much time on our pbx, but this is also where we test our patches to asterisk, so it can't be called against asterisk. My execution of this setup hasn't been telco quality, but seems pretty on par with small office pbx systems. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snatching calls
On Wed, 2003-11-05 at 02:32, Shoval Tom wrote: It works with SIP and with zap channels. What about IAX? like DIAX softphone? I may be misunderstanding something. When you start an Asterisk configuration process, connecting your hardware and building your dialplan, you use Zapata.conf, sip.con and iax.conf to connect the FXSs and FXOs to your Asterisk, as you would connect your normal lines and extensions to a regular PBX. The extensions.conf is used not only to devise the Dial plan, but also to assign extension numbers to all the extensions connected (be it sip, iax, zap, or whatever). Now I get to the groups. Call groups. Pickup groups, whatever groups. Isn't it logical that it would be done in extensions.conf? or in an other single location, and not throughout several files that you have to recheck every time you want to change anything? No it isn't logical to put it in another file or in extensions.conf. extensions.conf is JUST your dialplan. You define your users via the channel configurations. Just as the question that started this thread, they want to make these definitions based on who is in a specific room. This is only possible when you define the access to the device just like you define the voicemailbox. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Liew Sent: Wednesday, November 05, 2003 6:10 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] snatching calls - Original Message - From: Billy Huddleston [EMAIL PROTECTED] how could you do this with sip and VOIP? From: Steven Critchfield [EMAIL PROTECTED] You want to look into call groups and pickup groups. To pickup the call you use *8#. from /usr/src/asterisk/configs/zapata.conf.sample ; ; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing ; and it is a member of a group which is one of your pickup groups, then ; you can answer it by picking up and dialing *8#. For simple offices, just ; make these both the same ; callgroup=1 pickupgroup=1 As per what Steve says, the same applies for SIP, check /usr/src/asterisk/configs/sip.conf.sample Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] one way sound with x-lite (sip) -second attempt
Hi all, Solution found. Asterisk CVS-10/29/03 is simply BAD for chan_capi and X-ten Lite. New version Asterisk CVS-11/04/03 does the job. Haven't done much testing but 2 way sound is there. So everybody using this version of asterisk go for a new cvs... Thanks, Thorsten -- Hi all, Still having the one way sound problem. Any suggestions how to hunt the problem down ? Regards, Thorsten --- Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26 chan_capi On the IP side: X-lite (build: 1084) Calling and get calls on PSTN from X-Lite is no problem. We only get sound from PSTN to X-lite. Never from X.-lite to PSTN. The soundmeter on X-lite shows activity ... (not muted, correct device...) When pressing numbers while having these silent calls in x-lite is playing DTMFs at the PSTN phone side. sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to allow=all [1*phonenumber*] type=friend username=NAME secret=testpass auth=md5 nat=no host=dynamic reinvite=no canreinvite=no dtmfmode=inband callerid=Test *phonenumber* context=sip-phone-out Any suggestions ? Thanks, Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best place to order Cisco ATA 186
I want to setup an Asterisk network using the Cisco ATA 186. What is the best place to order those devices? I'm not finding them anywhere. Al __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] First AGI help..
Paul Liew wrote: Hi ?php // From Kapjod's sample.. ob_implicit_flush(true); set_time_limit(0); $err = fopen(php://stderr,w); $in = fopen(php://stdin,r); $out = fopen(php://stdout,w); //This works.. fputs($out, Verbose \Calling phone\n); // This doesn't fputs($out, exec(Dial(sip/2012)\n); fclose($in); fclose($out); fclose($err); ? You'll find its to do with your syntax - show agi exec produces Usage: EXEC application options Executes application with given options. Returns whatever the application returns, or -2 on failure to find application ie use spaces not (. Paul ___ Thanks Paul.. I was trying to glue bits from other scripts together and I guess I misinterpreted the syntax from the other scripts.. So thats working now I can try more interesting things and hopefully get somewhere in learning AGI.. Thanks again.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi, - Original Message - From: Shoval Tom [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 1:34 PM Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform) Dan, I can't seem to transfer calls using #. How is it supposed to be done? In the dial line (from extensions.conf) you must put t and/or T as the last parameter. Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best place to order Cisco ATA 186
On Wed, 2003-11-05 at 12:36, Al wrote: I want to setup an Asterisk network using the Cisco ATA 186. What is the best place to order those devices? I can't speak for the best place... but I found this: http://www.pricegrabber.com/search_getprod.php/masterid=614321 However, given that this cost is multiplied by the number of phones... do keep in mind a channel bank solution, since aside from the channel bank (anywhere from $100 - $500 on eBay), you will only need one of Digium's T100P cards ($495) to provide support for up to 24 analogue phones (or mix and match the 24 between FXO and FXS with the right cards for the channel bank) Significant savings to all but the smallest solution (a Digium TDM400P might be a better solution) Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent Logoff question
Hello there, Does anyone know how an agent logged on with AgentCallbackLogin application can logoff ? Thanks, Anna Anna Panagidou Technology Department Hellas On Line Agiou Konstantinou 59-61 15124, Maroussi Tel. no: (+30210) 8762309 E-mail address: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outband DTMF on i4l modem
Best of luck on getting an answer, I have posted several times with the same question. Unfortunately my time to reverse engineer this problem right now is low but my temporary solution's cons are pushing me to jump into the code and fix the problem. As a workaround you can set your Cisco phones to disable out-of-band DTMF and on Asterisk set the sip.conf entries for these phones to use inband DTMF. You will be able to get external IVR to work but you will probably have to use something like G.711 so the quality is hight enough. The codec usage depends on your configuration and you may be able to get away with some compression. I currently have some Cisco 7960's for phones and an Audiocodes 4-port FXO gateway installed. When using out-of-band DTMF, Asterisk DTMF functions worked fine. I could not hear DTMF between the Cisco 7960s (who really cares). The real problem was that I could not get DTMF relayed from the Cisco 7960 to PSTN via Asterisk and the Audiocodes. This is where I have to sit down with a packet sniffer and debug logs to see what is going on with the RTP. I need to know if the problem is in Asterisk relaying the phone events containing DTMF or if it is the Audiocodes not generating DTMF tones on the analog side. Hope this rambling helps. Maybe it will prompt someone else to chime in with a solution. :) Matthew Enger wrote: Hello, I am setting up 2 ISDN 4 linux cards and have had great success now that I have got over the initial problems with : and / characters. The only problem I am experiencing now is the sending of DTMF tones over the line to a remote IVR system. If I dial SIP (Cisco 7905 and 7940) to a number over the line, no DTMF tones are heard. I dialed my own home phone and tried it, no matter which button i pressed, no tones came out. However when I dial voicemail, the buttons work fine. Is this a problem with asterisk not putting the tones onto i4l? Thanks, Matthew Enger [EMAIL PROTECTED] Xintegration ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g.729 codec registration
Hi i'am again... i have tesed if my * (where the purch. g729 is installed) take calls from a gateway with g.729A codec. The calling mechanism works but there is no voice only bad noises . I'am a little bit confused. On the digium site i bought a g729 codec (without any indication of an a or a b). I thought this codec could take calls with g729.a codec but this seems not to be so. If my fiction is right, how can i take calls with g.729.a codec ? Thanks, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Mittwoch, 5. November 2003 12:15 An: Asterisk User Betreff: [Asterisk-Users] g.729 codec registration Hi all, i have purchased the g.729 codec from digium. The registration was successful. (with the old binary) But there're a few questions: - should not the codec listed in the codec list when i enter show codecs ? - the codec is named with g729b but if i enter show codecs there is a codec g729a listed also the g729b is not installed. what is the difference between g729a built in * and the puchased g729b codec? Thanks for help. Regards, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [REPOST] [Asterisk-Users] ZapRAS docs needed...
Steven Critchfield wrote: On Wed, 2003-11-05 at 04:55, Roy Sigurd Karlsbakk wrote: Is there any information available about ZapRAS other than the fscking source? just pointing out for those interested in reading that I have written up my recent experience over on the -dev list where I read this first. Ollie, you want to pick it up and find a suitable place for it in the wiki? Done. http://www.voip-info.org/tiki-index.php?page=Asterisk+zapras /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP with CIC
Hi, Is there any (easy) way to get Asterisk to include CIC-information in the SIP INVITE? CIC: http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/sip_c/sipc1_c/chapter3.htm#1314580 I need my SIP INVITE to look something like: INVITE sip:5550001;[EMAIL PROTECTED]:5060 SIP/2.0 I'v tried a couple of different things but can't find anything that works. I sure hope that there is another way besides diving into the sip channel source code... :) Regards, Niclas Gustafsson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Missed calls/activity log in Asterisk
I wonder what would be the easiest way to con Asterisk into logging all activityon ISDN line? Likeincoming calls, outgoingetc, even if these calls did not originate/terminate at Asterisk server? I'm using chan_capi if that matters (it should), with Fritz PCI S-type ISDN connection. TIA, Peter
RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
I've had t a the end, I've added T, although this is not necessary. Still doesn't work, though When calling or called party press #, nothing happens. Asterisk's console doesn't show anything, either. Can you send me a sample of an extension definition that works? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Sent: Wednesday, November 05, 2003 3:45 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Hi, - Original Message - From: Shoval Tom [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 1:34 PM Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform) Dan, I can't seem to transfer calls using #. How is it supposed to be done? In the dial line (from extensions.conf) you must put t and/or T as the last parameter. Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi, - Original Message - From: Shoval Tom [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 2:32 PM Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform) I've had t a the end, I've added T, although this is not necessary. Still doesn't work, though When calling or called party press #, nothing happens. Asterisk's console doesn't show anything, either. You must see in the console something like that: -- Playing 'pbx-transfer' A standard dial is : exten = 123,1,Dial(IAX/user1,30,tTr) then the user1 (on DIAX) can trasfer the call using '#' key. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does IAX pass ISDN result codes?
At 08:34 PM 11/4/2003 -0600, you wrote: Quoting Chris Ziomkowski [EMAIL PROTECTED]: I can't try this setup yet (still don't have the hardware), and have been trying to answer this question merely from the source code. So far, I have not been able to convince myself. Does anyone have definitive information on this? Passing q931 disconnect causes doesnt work on iax. rgrds m. Lovely. Yet another thing I'll have to hack up before I can get this project off the ground. It's a good thing asterisk is open source. Thanks for the information Martin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Brian, I think some of the confusion comes from what end of the line we are looking at and the nature of the imbalance. While the resistor may fix the near end, it will probably cause some termination problems at the far end. Reflections mostly, which on a short run analog line shouldn't be much of a problem. I haven't looked into this in detail. Also, if you look at the structure of a balanced transmission line, it should be really important to not have any imbalance out on the distributed part of the line, such as caused by having one wire of the pair having a different resistance, or by having a resistance anywhere but at the line termination - say 2000 feet out. If I interpret things correctly, this would give a line which has two termination resistances at which there is a peak of power transfer to the load, and neither of them would appear purely resistive, giving phase shift errors which make balancing the hybrid difficult or impossible, and degrading data transmission capability. Re the gain specs, I don't have a reference to them, but I suspect that there is stuff to be found on Google. Brian D Heaton wrote: Stephen, Very interesting. I know I've seen all manner of messiness in the past when folks have monkeyed with balanced pairs. I'll take your word on the modeling data. I've not gone that far in depth with it. You don't have the specs on gain adjustment handy do you? I've probably got it buried in an old pub somewhere, but I don't have anything in soft-copy. THX/BDH On Tue, 2003-11-04 at 18:34, Stephen R. Besch wrote: I just finished modelling a standard 4-transformer hybrid coupled to a balanced RC transmission line. Cross talk was zero when the hybrid was balanced. Inserting a single resistor in series with tip or ring imbalanced the hybrid and cross talk appeared. This could be completely compensated with the proper RC on the opposite side of the hybrid, as predicted. It made absolutely no difference to the cancellation if the resistor was split. Since a balanced hybrid appears as a pure resistance (complex terms are 0) to the transmission line, placing a simple resistor in series with the hybrid (on either side) at the termination point will just look like 2 resistors in series and will properly terminate the line. There should be no effects at all from doing this other than the loss of some energy in the termination resistor, which can be made up for with a boost in Rx gain. That's because the cores saturate on transformer based hybrids. This is not as likely to occur with active hybrids built with op-amps (which are found in almost all modern line cards), although it is possible if the gains are high enough. However the distortion from the clipping would be far worse than the echo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX clients and the flash button
-- Original Message -- From: Dan [EMAIL PROTECTED] Hi, Hi guys As usual I am playing around with IAX soft clients. I was wondering with the various IAX clients, IAX client, DIAX, etc how's one park calls, transfer calls, etc since there is no flash key? With DIAX you can transfer using '#' key. Then transfer the call to 701 Your right that you can transfer with the # key. But due to allot of systems needing the # we have disabled it here in our Asterisk system. The # key is used for pagers and other calls. That is why we would like to get a flash key! By the way the sound is better with DIAX then with Xten lite! Great job can't wait for your updates! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323: New version 0.5.6
Hello all, A newer version of asterisk-oh323 is available. This version features a set of channel variables and improvements in audio frame handling. People that have reported clicks or choppy sound, in some cases, should try this version. Download from: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX clients and the flash button
Hi, .. But due to allot of systems needing the # we have disabled it here in our Asterisk system. The # key is used for pagers and other calls. That is why we would like to get a flash key! I don't think is implemented in the library..:( By the way the sound is better with DIAX then with Xten lite! Great job can't wait for your updates! This is not my merit... I just use the library provided by Mark...:-) Anyway, I wait for your feedback in order to solve as many bugs as possible in the next release. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best place to order Cisco ATA 186
Hello Al, Please let me know how many of them do you want to order, we are distributors of Cisco and can help you . Als owe are providing International/Domestic calls termination to more then 260 countries worldwide. Thanks, Alexander Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: Al [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 7:36 AM Subject: [Asterisk-Users] Best place to order Cisco ATA 186 I want to setup an Asterisk network using the Cisco ATA 186. What is the best place to order those devices? I'm not finding them anywhere. Al __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Jorge Mendoza wrote: I'm in agree with all explanations regarding the echo and 2/4 wires conversion. However I'm wondering if there are other parameters like CPU and/or Asterisk configuration involved in the problem with more weight than hybrid. Otherwise how do you explain the difference in the following scenario: 1.- Crystal clear voice: [phone1][pabx]-[fxo gateway]--SIP---[fxs gateway]--[phone2] 2.- A lot of echo: [gnophone or xten]---[ * ]---SIP-[fxs gateway]-[phone2] First, I assume that the right hand side in both cases is the same. Then determine where the echo originates. I would guess that it is either coming from the fxs gateway and that there is no echo cancellation running on the fxs, while there is on the fxo, or you are using an open mic and speaker on the softphone. In the first case, when the fxo is in the circuit, the echo canceller sees the echo - it doesn't care where it comes from, the signal will stii autocorrelate because it contains a copy of itself - and it gets removed. In the other case, the echo does not get removed because the echo canceller is not running on that channel. Examine the channel status (zap show channel x) and the canceller is active when a call is in progress. In the second case, use a headset. Echo cancellers are notoriously poor at cancelling room echo very well without very compute intensive algorithms and long tail lengths (lots of taps). It usually reguires a DSP. If it's neither of these, then I am stumped. The first scenario has four 2/4 W conversion. Well, I only see two possible hybrids here, unless the PABX is running analog at the external ports and digital internally -nutty, but possible. The second one has only two (or one?). Probably one. The * was running in differents CPU, PIII 500 Mhz, PIII 750 Mhz, 128 Mb to 512 Mb ram with not difference on echo. We have installed a Mitel 3100 with IP phones at 40 kms within a wireless network with not echo at all. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need info on Gastman/Astman
Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon. Services like Conference bridge and Musichonhold seem to work ok (I use [EMAIL PROTECTED] and [EMAIL PROTECTED]) for the Icon extensions. IAX softphone seems to work ok (I use IAX/[EMAIL PROTECTED]) for the Icon extension But for SIP phones, I use SIP/311 for an extension. But when the phone is used (either dialing out or being dialed to) a new icon pops up on the screen (SIP/311-ferh). If you have a working Gastaman, can you share your configuration file , please? Anyone have any documentation on Gastman/Astman? Thanks Lee Goodman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reasons why I shouldn't use Asterisk?
It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' Can anyone think of any others? Cheeres, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need info on Gastman/Astman
Lee Goodman wrote: Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon. Services like Conference bridge and Musichonhold seem to work ok (I use [EMAIL PROTECTED] and [EMAIL PROTECTED]) for the Icon extensions. IAX softphone seems to work ok (I use IAX/[EMAIL PROTECTED]) for the Icon extension But for SIP phones, I use SIP/311 for an extension. But when the phone is used (either dialing out or being dialed to) a new icon pops up on the screen (SIP/311-ferh). If you have a working Gastaman, can you share your configuration file , please? Anyone have any documentation on Gastman/Astman? Thanks Lee Goodman This question has come up before.. IIRC its to do with the fact that a SIP call has a uniqe ID apended to it on each call so it doen not play nicely in Gastman.. I guess that the Gastman code should be modified to strip off the unique ID from the SIP channel reference.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
On Wed, 2003-11-05 at 09:08, Gavin Hamill wrote: It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' Your listed con is only a con if you have to point to others failures to cover your own. It is lemming thinking. Can anyone think of any others? If your company has a few decent programmers with a good knowledge of open source software and software debugging, the few access of the source code is the greatest PRO. I think the only Con we could think up when we initially deployed was our question of the longevity of the Digium company. Every time I get to hear from Mark about how the company is doing, the less this concern becomes. This was only a concern because of the need to have affordable and well supported channels into asterisk. We choose to download all the software at that point and all the schematics and information, burn it to CD, and store it away in our lock box as a safety net. I'd say the only other cons you could list are really in driver support areas right now, but knowing that those are moving targets and can potentially be fixed or avoided lowers those risks. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
Gavin Hamill wrote: It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' Can anyone think of any others? No built in high availability or clustering options making it as reliable as the harware, OS and apps.. Last time I looked it up PC systems combined hardware components average reliability was about 96% uptime(This was a while back so the percentage may not be accurate).. This is a problem for telecom's system whos uptime is usually measured in years and not a percentage of 1 year.. No flames please, I realise that there are issues involved with the PSTN lines, channel banks and some other things in a clustered senario.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX clients and the flash button
On Wed, 2003-11-05 at 08:32, Dan wrote: Hi, .. But due to allot of systems needing the # we have disabled it here in our Asterisk system. The # key is used for pagers and other calls. That is why we would like to get a flash key! I don't think is implemented in the library..:( By the way the sound is better with DIAX then with Xten lite! Great job can't wait for your updates! This is not my merit... I just use the library provided by Mark...:-) Anyway, I wait for your feedback in order to solve as many bugs as possible in the next release. Dan, Ariel is asking you to attack this problem incorrectly. You are correct that there doesn't seem to be a flash function in IAX. There doesn't need to be a flash function as you can signal out of band. Specifically for transfer there is these IAX_FRAME_IAX subclasses; #define IAX_COMMAND_TXREQ 22 /* Transfer Request */ #define IAX_COMMAND_TXCNT 23 /* Transfer Connect */ #define IAX_COMMAND_TXACC 24 /* Transfer Accepted */ #define IAX_COMMAND_TXREADY 25 /* Transfer ready */ #define IAX_COMMAND_TXREL 26 /* Transfer release */ #define IAX_COMMAND_TXREJ 27 /* Transfer reject */ This means you just need to make a software transfer button. Of course while your at it, you might want to look into the three way calling functions, and any other calling functions that might need to be added that would otherwise be used by flash hooking a zap channel. Hope this helps. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't connect voice in Zplex 10B
Hi Everybody I'm connecting a zplex 10B to a TE410P card on my asterisk server (i use a cross cable), the zplex doesn't show any alarm neither the server. I'm trying to make calls from 1 line from the channel bank, to another, but when I dial the ext number, the dialtone doesn't stop... every way i make the call and the called extension rings but when i pick up the phone i just listen the dialtone. I programmed the lines in the zplex as fxs and i made a voice cross connection. I don't known what to do, for been able to connect the voice, any idea? Thanks for yours answers Att Yelson Vivas . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323: New version 0.5.6
Michael Manousos wrote: A newer version of asterisk-oh323 is available. This version features a set of channel variables and improvements in audio frame handling. People that have reported clicks or choppy sound, in some cases, should try this version. Download from: http://www.inaccessnetworks.com/projects/asterisk-oh323 What's the difference between this package and NuFone's implementation in the CVS? /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
Gavin Hamill wrote: It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' Can anyone think of any others? Cons: * Not a full SIP proxy If you're looking for a SIP proxy that follows the RFC, Asterisk is not your choice. ...yet. There's work going on to fix this. * No release handling There are new schemes planned for stable/development branches but right now, there is only your pick of a CVS date and you're on your own to see if it's stable. New functions aren't downported to older, stable, versions. * Limited hardware support The software is pretty well tied to Digium hardware for PSTN connectivity. (Myself, I have no problem with this, but it could be seen as a con). /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using Asterisk as a VOIP gateway
Is it possible to use * as a VOIP gateway? Can I connect asterisk to one of the trunks on my current PBX and on the other side of the world connect another * to the trunk of another regular PBX is it possible to transfer calls from here to there? I guess I'll need one port FXO card for each asterisk, but I can't figure how to configure the thing. I know I'll need to configure the regular PBX to forward certain calls to the lines connected to asterisk (by prefix, or just have everyone dial 8 and get a line) Does this scenario make sense to anyone? Or am I barking up the wrong tree? Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: [EMAIL PROTECTED] Mobile: 972-55-229220
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
On Wed, 2003-11-05 at 15:31, Steven Critchfield wrote: the only Con I can think of is 'Relatively few installations worldwide' Your listed con is only a con if you have to point to others failures to cover your own. It is lemming thinking. Hm, it's only a con because I can't think of anything else, to be honest :) The call centre is our lifeblood at the moment, so whilst I personally am confident that Asterisk can perform the task more reliably and more adaptably than our current PBX, the fact that not many other people (that I know of) are using the software is a significant factor in the risk assessment. Our mgmt are reasonably open and forward, but with something as core to the business as the phones, I expect they will show a more conservative side. It's a pleasant boost to my case that our current proprietary PBX has been dogged with problems. Can anyone think of any others? If your company has a few decent programmers with a good knowledge of open source software and software debugging, the few access of the source code is the greatest PRO. Alas, we're a web shop, so the coding talent extends only to Perl/PHP. Perfect for AGI scripting, but not a lot of use if we find issues in the Asterisk core. I think the only Con we could think up when we initially deployed was our question of the longevity of the Digium company. Interesting - something I'd not even considered simply because new products appear all the time, and I haven't heard a bad word said about them to date. I'd say the only other cons you could list are really in driver support areas right now, but knowing that those are moving targets and can potentially be fixed or avoided lowers those risks. nod Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
On Wed, 2003-11-05 at 09:36, WipeOut wrote: Gavin Hamill wrote: It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' Can anyone think of any others? No built in high availability or clustering options making it as reliable as the harware, OS and apps.. Last time I looked it up PC systems combined hardware components average reliability was about 96% uptime(This was a while back so the percentage may not be accurate).. This is a problem for telecom's system whos uptime is usually measured in years and not a percentage of 1 year.. No flames please, I realise that there are issues involved with the PSTN lines, channel banks and some other things in a clustered senario.. I think the number you cited needs qualification to be accurate. Because if it where accurate as it stands, I'm due for major downtime in my rack as I have several systems approaching 2 years uptime without a single hardware failure. These machines also where not new when they where sent to the colo facility. In fact they all had been running for about a year before hand. And as a question of the 5 9's reported on telco hardware, As far as I know, that is for total system failure. The fact that they could loose trunks, or even a portion of a neighbor hood doesn't count against their downtime. If it did, I could point to a couple of telcos in this area that would have problems meeting those requirements. --- to back up my claim about uptime, my webserver is showing 136 days uptime, this is after a 497 day wrap around of the uptime counter. This machine is a Dell pe2450 the mail server is a home built 700 celeron showing the same 136 day uptime after the 497 day uptime wrap around. Due to a hacker, our clients machine is showing 105 days uptime post 497 day uptime wrap around. Again home built machine. One of our fileservers is showing 133 days uptime post uptime wrap around. This is due to a screw up at the keyboard just 3 days after installing it in the colo. Also a home built machine. Our VPN machine is just getting up to 354 days uptime. This is a super micro we purchased and put into service shortly there after. Our database server just went through a hardware and software upgrade that caused it's reboot, now at 185 days uptime. Same hardware as the above listed webserver. The 2 machines in my rack without impressive uptimes are a NT machine and my phone gateway that just had a kernel update. This should probe that good power supply to the machine will help make hardware run well for a long time. Why do you think the telco equipment runs on 48volts? They are pulling from the batteries 100% of the time. This makes a smooth even power flow. Machines in my office are subjected to poorer quality power and tweaking so they don't tend to make it to the 200 day uptime mark very often. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Demo Weather Report AGI v2.0
Cepestral was installed and working within 10 mins of my decision to purchase it. It's $30.00 and can be purchased on their web site and they give you a download. They have a demp on their website that will do text-to-speech and give you a .wav file to download and listen to. Download, unpack it, run their install.sh, answer a couple of questions, read the man page and you're done. With Festival I had to figure out exactly which tarballs to download (there was a total of 18 tarballs to download if you count all the Festival voices plus the MBROLA voices), then I had to figure out how to install Festival, then MBROLA, I never have figured out how to actually INSTALL festival, I just run it out of the source directory. It's very picky about paths and such. I'm not a big fan of commercial software. For TTS most of the software either is Windows only or costs several thousand dollars (and sometimes both). If it's a choice between spending two thousand for something like Rhetorical TTS or using Festival, I'll pick Festival. If it's a choice between spending thirty dollars for a TTS system or using Festival, I'll happily spend the $30. Thats a very easy ROI since one hour of a technical resource to setup Festival is easily double the 30 USD. Maybe the Cepestral folks have figured out that making a little money from alot of people will be much better than alot from only a few. I'll buy Cepestral and skip the pizza on Friday night. Net result will be about break even ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remote Call Pickup
Hello, Can somebody point me where I can find examples how I can set up Remote Call Pickup ? Thx Bart
Re: [Asterisk-Users] IAX clients and the flash button
On Wed, 05 Nov 2003 09:36:28 -0600, Steven Critchfield wrote: On Wed, 2003-11-05 at 08:32, Dan wrote: Hi, .. But due to allot of systems needing the # we have disabled it here in our Asterisk system. The # key is used for pagers and other calls. That is why we would like to get a flash key! Might I make a suggestion here Most win users, who are in a residential enviroment can get terribly confused with multiline phones and such... Whilst thigs like call transfer, call waiting, dirversion and such is becoming much more common, many users really just don't need it. So if at all possible, how about a default option of the program being just a single line device, with the option of turning the extra's on etc. I would luv to be able to distribute DIAX (or any program similar) as a self extracting archive where we have already added a basic config file for the individual user so they really don't need any knowledge what so ever to run it... its just a plug in an go operation for them. Having said that, it would also be nice to have some sort of interface via a web page which they could access within the program (being an individual page based upon their login details. over remember the KISS principle please. Gary ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
One of the biggest cons is the lack of friendly interface for configuration. However, most PBXs in use don't have one either, unless they are about 5 years old or newer, in which case it probably wouldn't be on the chopping block. I still think the pros outweigh the cons, or else I wouldn't be on this list :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Hamill Sent: Wednesday, November 05, 2003 9:08 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Reasons why I shouldn't use Asterisk? It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' Can anyone think of any others? Cheeres, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
On Wed, 2003-11-05 at 16:02, Olle E. Johansson wrote: Cons: * Not a full SIP proxy Fortunately this is not relevant to our environment :) * No release handling Good point, I've added that to the list.. * Limited hardware support The software is pretty well tied to Digium hardware for PSTN connectivity. I don't see this as an issue, either :) Digium's hardware is expensive only for those in a computing environment where even complex hardware costs only a few dollars... From a telco pricing view, $1495 for a card that lets you connect four PRIs is a bargain :) Many thanks for your suggestions. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323: New version 0.5.6
Olle E. Johansson wrote: Michael Manousos wrote: A newer version of asterisk-oh323 is available. This version features a set of channel variables and improvements in audio frame handling. People that have reported clicks or choppy sound, in some cases, should try this version. Download from: http://www.inaccessnetworks.com/projects/asterisk-oh323 What's the difference between this package and NuFone's implementation in the CVS? asterisk-oh323 was the first H.323 channel driver for Asterisk. The other, included in CVS, is based on this one, but it uses the RTP stack that comes with Asterisk (when we first wrote this software, there was no RTP stack in Asterisk). There are also some other differences (in features, performance, stability), but I think that a third person could make a more unbiased list. /Olle Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Phone Review: Clipcomm 101
Hello, I have received yet another new phone today, the ClipComm 101 (http://www.clipcomm.co.kr/eng/e_product/e_product_voip_ip_phone.html) I bought it for $165 directly from the Korean Manufacturer(No US distributer yet). Here are the features: - Built-in NAT functionality, you can switch from Hub to Nat, great for home DSL/Cable users - This includes some limited port forwarding functionality, very cool idea(wish GS had that) - PSTN port included, it can be hooked up to both VOIP and a POTS line - Full duplex Speakerphone - large LCD display - SIP compatible, but I could only place outgoing calls through a proxy, not directly through Asterisk, incoming calls from any asterisk extension went through just fine. I couldn't disable proxy and use a local server for some reason, but it has old firmware, so maybe after I upgrade it. - Phonebook and call log - they have a wireless bluetooth version too(I didn't get that one though) - 5 second reboot Conclusion, very cool phone, wish proxy could be disabled, but this phone is really aimed at the home market. NAT/Router functionality is a great idea, makes NAT traversal on phones a non-issue for home users. The price is good but no US distributor means shipping is steep($30 for mine), take a look at the specs on their site, your be impressed what you get for the price. MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] get IP Address from caller using oh323
Thomas Haeger wrote: Hi all (Michael), how it is possible to get the ip address of the calling party ? (i know by using h323... but there're a few unknown segfaults...) and so i want to use oh323, but i have to get the ip from the caller to permit or deny the call with AGI. Is it possible at all ? Yes. Try the new version (0.5.6). The IP of the calling party is available inside a couple of variables maintained by the channel driver (check the README). Thanks, Thomas. Michael. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi, Sorry to answer just now. your mail was lost between others from the same list..;) - Original Message - From: Asterisk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 10:37 AM Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform) Hello Dan, Its an excellent start. Please don't get swayed away by some stupid remarks. I am really impressed by your work and I hope to see a lot more releases from you. I'll keep it up. In spirit of improving the code, here are some of the issues that I faced while trying it out: 1. Once I dial the number, the directory disappears and never shows up What do you meen by directory? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Asterisk as a VOIP gateway
yes you can but may not be all that economical though. on the other hand, if you can replace or do away with at least one of the pbx with * at either end, i think you'll be ahead of the game :-) Shoval Tomer wrote: Is it possible to use * as a VOIP gateway? Can I connect asterisk to one of the trunks on my current PBX and on the other side of the world connect another * to the trunk of another regular PBX - is it possible to transfer calls from here to there? I guess I'll need one port FXO card for each asterisk, but I can't figure how to configure the thing. I know I'll need to configure the regular PBX to forward certain calls to the lines connected to asterisk (by prefix, or just have everyone dial 8 and get a line) Does this scenario make sense to anyone? Or am I barking up the wrong tree? Shoval Tomer , MCSE IT Manager Softov Advanced System Ltd. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Mobile : 972-55-229220 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
--- Gavin Hamill [EMAIL PROTECTED] wrote: It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' What's few there must be many thousands of installations. Reasons not to buy it, I think all revolve around the fact that Digium is a very small bussiness and could easly vanish. and while there are numerus small consulting firms that could step in few have to ability to actually write new core PBX code or device drivers and continue development. Even those few might see that someone motivated and talented failed and not want to step in. That said, if you have some technical skills you don't need support and if all else fails there are other Open Source projects that you could fall back on. So the risk is quite low. It is _very_ low if you take the time to make plans to cover yourself. In the end, you have the Asterisk code, you don't get this if you buy a comercial PBX system. The other reason not to buy is that it simply may not be a good technical fit. Clearly Asterisk is not what you'd want if your company had 10,000 phone extensions Can anyone think of any others? Cheeres, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
On Wed, 2003-11-05 at 10:16, Gavin Hamill wrote: On Wed, 2003-11-05 at 15:31, Steven Critchfield wrote: the only Con I can think of is 'Relatively few installations worldwide' Your listed con is only a con if you have to point to others failures to cover your own. It is lemming thinking. Hm, it's only a con because I can't think of anything else, to be honest :) The call centre is our lifeblood at the moment, so whilst I personally am confident that Asterisk can perform the task more reliably and more adaptably than our current PBX, the fact that not many other people (that I know of) are using the software is a significant factor in the risk assessment. Our mgmt are reasonably open and forward, but with something as core to the business as the phones, I expect they will show a more conservative side. It's a pleasant boost to my case that our current proprietary PBX has been dogged with problems. Can anyone think of any others? If your company has a few decent programmers with a good knowledge of open source software and software debugging, the few access of the source code is the greatest PRO. Alas, we're a web shop, so the coding talent extends only to Perl/PHP. Perfect for AGI scripting, but not a lot of use if we find issues in the Asterisk core. You would be surprised at how quickly you can pick up on how things are done in a well coded C project. I assume if you are doing php/perl, you probably are at least passingly familiar with a linux system and can pick up some easy debugging skills. I think the only Con we could think up when we initially deployed was our question of the longevity of the Digium company. Interesting - something I'd not even considered simply because new products appear all the time, and I haven't heard a bad word said about them to date. Our concerns where only about them being a small company and you have to understand our implementation is over a year old. I think we purchased parts before the official FCC certs where in. This all said, we where cutting it close to being on the very cutting edge of what was released. You are now coming into it after Digium has grown and released a few more products. We no longer have these kinds of fears, but 18-24 months ago would have been a little different. Another reason why we may have thought about it was simply the fact that we also are a really small shop that occasionally have to worry about how business will fair. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX clients and the flash button
Hi Steven, - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 5:36 PM Subject: Re: [Asterisk-Users] IAX clients and the flash button ... Dan, Ariel is asking you to attack this problem incorrectly. You are correct that there doesn't seem to be a flash function in IAX. There doesn't need to be a flash function as you can signal out of band. Specifically for transfer there is these IAX_FRAME_IAX subclasses; #define IAX_COMMAND_TXREQ 22 /* Transfer Request */ #define IAX_COMMAND_TXCNT 23 /* Transfer Connect */ #define IAX_COMMAND_TXACC 24 /* Transfer Accepted */ #define IAX_COMMAND_TXREADY 25 /* Transfer ready */ #define IAX_COMMAND_TXREL 26 /* Transfer release */ #define IAX_COMMAND_TXREJ 27 /* Transfer reject */ This means you just need to make a software transfer button. Of course while your at it, you might want to look into the three way calling functions, and any other calling functions that might need to be added that would otherwise be used by flash hooking a zap channel. You're right. I intend to add three way calling and transfer button only after IAX2 will be the default protocol of the application. I work on it. Hope this helps. A lot. Thanks. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Where can i get the g.723 codec?
Hi Andrew, Tuesday, November 4, 2003, 5:04:04 AM, you wrote: AJ I have used G723.1 (although unlicensed) with Asterisk. The info is even AJ in the Makefile, just drop in a few files in your source directoy, AJ uncomment something in the Makefile and instant G723.1 support... Thanks for the info, but after looking at the Makefile, I am not sure where I can get the source code for G723? Can you provide a hint? TIA -- Best regards, Nguyenmailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need info on Gastman/Astman
On Wednesday 05 November 2003 08:47, Lee Goodman wrote: Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon. Services like Conference bridge and Musichonhold seem to work ok (I use [EMAIL PROTECTED] and [EMAIL PROTECTED]) for the Icon extensions. IAX softphone seems to work ok (I use IAX/[EMAIL PROTECTED]) for the Icon extension But for SIP phones, I use SIP/311 for an extension. But when the phone is used (either dialing out or being dialed to) a new icon pops up on the screen (SIP/311-ferh). Yep, Zap channels work the same way. This is due to the fact that more than one call may be handled by the device at the same time. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
Yes, I agree. Your typical PC might have only 96% uptime but you could still build a __system__ with five nines of uptime using PC hardware. You eed to do two things. 1) Use better quality PC hardware that employs some internal redundancy, like mirrored drives and multiple load sharing power supplies. 2) design the system so that critical functions can fail over or at worst be restored quickly. Doing all this will triple (at least) your costs but that's just what it takes if you _really_ need all five of those nines. Yes it would be nice if someone could port Asterisk to Sun SPARC hardware then it could run on Sun's telco-grade Netra boxes --- Steven Critchfield [EMAIL PROTECTED] wrote: On Wed, 2003-11-05 at 09:36, WipeOut wrote: Gavin Hamill wrote: It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' Can anyone think of any others? No built in high availability or clustering options making it as reliable as the harware, OS and apps.. Last time I looked it up PC systems combined hardware components average reliability was about 96% uptime(This was a while back so the percentage may not be accurate).. This is a problem for telecom's system whos uptime is usually measured in years and not a percentage of 1 year.. No flames please, I realise that there are issues involved with the PSTN lines, channel banks and some other things in a clustered senario.. I think the number you cited needs qualification to be accurate. Because if it where accurate as it stands, I'm due for major downtime in my rack as I have several systems approaching 2 years uptime without a single hardware failure. These machines also where not new when they where sent to the colo facility. In fact they all had been running for about a year before hand. And as a question of the 5 9's reported on telco hardware, As far as I know, that is for total system failure. The fact that they could loose trunks, or even a portion of a neighbor hood doesn't count against their downtime. If it did, I could point to a couple of telcos in this area that would have problems meeting those requirements. --- to back up my claim about uptime, my webserver is showing 136 days uptime, this is after a 497 day wrap around of the uptime counter. This machine is a Dell pe2450 the mail server is a home built 700 celeron showing the same 136 day uptime after the 497 day uptime wrap around. Due to a hacker, our clients machine is showing 105 days uptime post 497 day uptime wrap around. Again home built machine. One of our fileservers is showing 133 days uptime post uptime wrap around. This is due to a screw up at the keyboard just 3 days after installing it in the colo. Also a home built machine. Our VPN machine is just getting up to 354 days uptime. This is a super micro we purchased and put into service shortly there after. Our database server just went through a hardware and software upgrade that caused it's reboot, now at 185 days uptime. Same hardware as the above listed webserver. The 2 machines in my rack without impressive uptimes are a NT machine and my phone gateway that just had a kernel update. This should probe that good power supply to the machine will help make hardware run well for a long time. Why do you think the telco equipment runs on 48volts? They are pulling from the batteries 100% of the time. This makes a smooth even power flow. Machines in my office are subjected to poorer quality power and tweaking so they don't tend to make it to the 200 day uptime mark very often. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
i m a newbie with * so in all likelihood my question will sound stupid to you but aren't there HA support for linux already? as to the pstn interfaces, i thought most traditional PBX uses redundant equipment to provide HA; can't we do the same with * being the switch? WipeOut wrote: Gavin Hamill wrote: It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' Can anyone think of any others? No built in high availability or clustering options making it as reliable as the harware, OS and apps.. Last time I looked it up PC systems combined hardware components average reliability was about 96% uptime(This was a while back so the percentage may not be accurate).. This is a problem for telecom's system whos uptime is usually measured in years and not a percentage of 1 year.. No flames please, I realise that there are issues involved with the PSTN lines, channel banks and some other things in a clustered senario.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
Steven Critchfield wrote: I think the number you cited needs qualification to be accurate. Because if it where accurate as it stands, I'm due for major downtime in my rack as I have several systems approaching 2 years uptime without a single hardware failure. These machines also where not new when they where sent to the colo facility. In fact they all had been running for about a year before hand. I agree.. Like I said those numbers were based on memory.. I researched it about a year ago for a customer I was consulting to.. Also I think the numbers were based on a population of PC's in a company and then converted to an average.. In any case I agree with you completely that systems are capable of running for a year or more uninterupted.. The fact still remains that CEO's and CFO's and any other board or management member seem to feel far more comfortable when a critical business system can be made as redundant and fault tolerent as is imaginably possible.. When you tell a person there is no OPTION for redundancy of the system they will tent to shy away and so that is why I said it was a potential con in the pro's and con's list.. And as a question of the 5 9's reported on telco hardware, As far as I know, that is for total system failure. The fact that they could loose trunks, or even a portion of a neighbor hood doesn't count against their downtime. If it did, I could point to a couple of telcos in this area that would have problems meeting those requirements. I agree with you here too.. 5 9's is alway a debatable statistic in the life of a system.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error in app_voicemail2.so after CVS update
Hi all, I have done some minutes ago a full CVS update, like that: cvs checkout zaptel zapata libpri asterisk cd zaptel make clean ; make install cd ../zapata make clean ; make install cd ../libpri make clean ; make install cd ../asterisk make clean ; make install When I try to start astersik with asterisk -vvc I get the following error and the program stops: [app_voicemail2.so]WARNING[1074412256]: File loader.c, Line 232 (ast_load_resource): /usr/lib/asterisk/modules/app_voicemail2.so: undefined symbol: ast_localtime WARNING[1074412256]: File loader.c, Line 400 (load_modules): Loading module app_voicemail2.so failed! if I try to add in modules.conf the line: noload = app_voicemail2.so I get the following error when starting *: [app_sayunixtime.so]WARNING[1074412256]: File loader.c, Line 232 (ast_load_resource): /usr/lib/asterisk/modules/app_sayunixtime.so: undefined symbol: ast_say_date_with_format WARNING[1074412256]: File loader.c, Line 400 (load_modules): Loading module app_sayunixtime.so failed! If I put: noload = app_sayunixtime.so then * starts ok, but I cannot use voicemail2 application. What can I do to solve this problem? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error compiling asterisk
I did cvs update on asterisk, zaptel, libpri as of today (November 5, 2003). I also did 'make clean' on each of them. My previous version of asterisk was cvs of September 15, 2003. No other changes have been made to my system other that these updates. when running 'make asterisk' the following error appears term.c:55: conflicting types for `term_color' include/asterisk/term.h:47: previous declaration of `term_color' term.c:98: conflicting types for `term_prompt' include/asterisk/term.h:49: previous declaration of `term_prompt' make: *** [term.o] Error 1 [EMAIL PROTECTED] asterisk]# Can anyone give me a hint of what the problem may be? Don Pobanz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web Interface for adding new users
Greetings to all, We are new to the Asterisk community and really appreciate thewillingness you have to share knowledge and help one another. We are building one of the Nations first fiber to the home networks here in Provo and have selected Asterisk to test as our phone switch. Newbie kinds of questions: Is there any web-based tool that would allow non-technical personnel to add new Telephone customers? Has anyone developed any database interfaces for billing data or Customer maintenance? Is everyone comfortable that Asterisk is capable of delivering five 9s, primary residential, 911 functioning service? Thanks for your input, Jeff
[Asterisk-Users] Skinny (SCCP) help
I have a cisco 7910 phone, I'm trying to get it to connect to asterisk, But it seems like it needs either a SEPDefault.cnf file or a SEPMACADDR.cnf file to Continue, I created empty ones but it's still sitting there saying opening Does anyone have examples of the SEPDefault.cnf file? Kevin, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] i4l-modem dtmf detection
hello! I have active call from i4l modem to ZAP (FXS).When someone on i4l (telco side) speaks i hear DTMF tones on other side (ZAP). How to turn off DTMF detection on modem-i4l side ? Is it possible to do that ?? status of active channels: server*CLI show channel Modem[i4l]/ttyi0 -- General -- Name: Modem[i4l]/ttyI0 Type: Modem UniqueID: 1068056585.53 Caller ID: 5 DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 64 WriteFormat: 64 ReadFormat: 64 1st File Descriptor: 8 Frames in: 10914 Frames out: 7514 Time to Hangup: 0 -- PBX -- Context: remote Extension: 0346546777 Priority: 2 Call Group: 0 Pickup Group: 0 Application: Dial Data: Zap/1/0346546777w||r Stack: 0 Blocking in: ast_waitfor_nandfds - server*CLI show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: 1068056588.54 Caller ID: 5 DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 68 WriteFormat: 64 ReadFormat: 64 1st File Descriptor: 18 Frames in: 5536 Frames out: 6378 Time to Hangup: 0 -- PBX -- Context: nme Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: Bridged Call Data: Modem[i4l]/ttyI0 Stack: -1 Blocking in: ast_waitfor_nandfds - server*CLI zap show channel 1 Channel: 1 File Descriptor: 18 Span: 1 Extension: Context: nmt Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: Zap/1-1 Real: Zap/1-1 (Linear) Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently ON Actual Confinfo: Num/0, Mode/0x Actual Confmute: No tnx. Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
Can anyone think of any others? mmh... some idea here * experienced linux user for production use (able to di compilation, knows how the shell works, able to debug code kernel probs, blah blah blah) * interoperating with other telco (even only lines...) needs some background in telecom world... like what I must do if my pri doesn't work ? I learned to debug pri messages, when a E1 of one customer didn't worked... also other issues... like echo or similar * must know how the net works... expecially in VoIP applications * again... a very experienced linux man to deploy robust reliable * servers ... just my 2 cents Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A real-life production scenario
Since it's all the craze, I might as well post our current Asterisk usage. :-) EQUIPMENT: - Beefyish box (dual Xeon 2.4GHz, gig of RAM, more-than-adequate disk space, etc) in a 1U chassis. - A second, slightly less beefyish box of specs I don't have handy right now, also in a 1U. - 2xTE410P CONNECTIONS: - 1 PRI to telco for local outbound/direct-dial inbound, 300 numbers attached. - 2 PRI to another telco for toll outbound/toll-free inbound - 1 EM T1 to office PBX We offer VoIP services to our directly-connected customers, ranging from simply taking their toll traffic to providing virtual PBX services, all using Asterisk. We've done a great variety of things (oddly, all customers are not alike)... here's a sampling: * Connection to our PBX Our PBX previously had a T1 in from a telco using an EM trunk, with 4 digits on the DNIS. When we had the Asterisk stuff stabilized, we wanted to move over to it ASAP because LD was much cheaper. (That, and the T1 wasn't the cheapest T1 we have here...) We disconnected one of the extra toll PRI's and, in its place, put the T1 from the telco. We then connected (using a crossover) the PBX to the TE410P. Various switching magic was performed (this was the point where I realized it's only getting 4 digits on the DNIS) and inbound calls were sent over to the PBX. Outbound calls from the PBX were switched like our VoIP calls. Following this, we ordered porting of that block of numbers over to the inbound PRI. The telco did it about 5pm on a Wednesday afternoon with no notification. Unfortunately, I had slightly bungled the exten = entry for calls coming in via that route. Fortunately, it was easy enough to fix, and was fixed before I got about the fourth swear word out of my mouth. The CDR file captured the caller ID on the confrangled calls, and our support department called them back promptly, and everyone was happy. * Customer with their own POTS lines wanting VoIP service One of our VoIP customers was in the interesting position of wanting the phone lines at their office, terminated analogly. We had a Mediatrix gateway in for testing, and decided to deploy it there. The Mediatrix was configured to send inbound calls to the Asterisk box, as well as gate 911 calls from the Asterisk to the PSTN (so that, when they call 911, it shows up with *their* location instead of *ours*). Calls from the Mediatrix successfully make it to Asterisk (with caller ID) where they ring the receptionist phone for 10 seconds then go to an auto-attendant/voicemail/etc. The Mediatrix doesn't answer (and therefore doesn't pass the call) until around the second ring, which is annoying, but them's the breaks. There's a bunch of other situations as well, but basically, it'll do most things. :-) -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What the Installation Instructions SHOULD HAVE SAID..
Hello all-- When I bought the TDM400P and the two FXO cards to prototype (small-scale) what could be done with Asterisk, I got a single sheet of paper with the cards, that explained how to insert the card, and fetch the source for asterisk, zaptel, and whatnot. But, before I could get it working, there were a few things that had to be done, that are not explained on the paper! So, to help the poor newbies who are starting from scratch, I humbly throw in the following two-cents, hoping to ease their pain. Please, feel free to correct anything that I have gotten wrong. First, the sheet I got is called Installation Instructions for the TDM400P (1-4 port FXS PCI Interface card). The Instructions are split into Hardware Installation, and Driver and Asterisk Installation. There is nothing wrong with either section. Actually, anything to add would be under a third section, Pulling it all together. Pulling It All Together Interrupt Priority. On Redhat Linux (and perhaps other brands), you can issue the command: cat /proc/interrupts and it will show you how the different hardware devices are assigned to the different interrupt levels or Iterrupt Request numbers, or whatever. Everything will work better if you have each card you just inserted on its own interrupt number. However, depending on your motherboard/BIOS combination, this may be difficult or impossible to achieve. You may swiftly find that the machine that will host phone hardware and Asterisk should be dedicated to this purpose, and all unnecessary options provided by the motherboard should be turned off in the BIOS -- at least, anything that requires an interrupt slot. For MY situation, everything fancy was turned off. USB taking up interrupt slots? Turn them off. Joystick, serial ports, etc? Disable them. The fewer things contending for interrupt slots, the better. Also, most BIOS setups allow you to assign interrupt numbers to cards, based on the slot you plug them into. Keep to the top slots in the system, closest to the AGP slot. Some have asserted that XT-PIC is an antiquated way to handle the interrupts, but I do not have the kernel option skills necessary to modify this so far. This is what I get: cat /proc/interrupts CPU0 0: 18311297 XT-PIC timer 1: 39583 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 182857694 XT-PIC wcfxo 5: 183672409 XT-PIC eth0, wcfxo 8: 1 XT-PIC rtc 10: 182826511 XT-PIC wcfxs 11: 45661576 XT-PIC es1371 12: 408175 XT-PIC PS/2 Mouse 14: 1688558 XT-PIC ide0 NMI: 0 ERR: 0 I personally have tried many different arrangements, and this seems the best I can do. I'd like to get the wcfxo card on a different interrupt number than the one that eth0 is on, but the system seems adamant about keeping them together. I'm using an MSI board, fairly late model. I have seen previous mailings, that seem to indicate that it is unwise to put more than 2 or 3 cards into a single machine. So, if you have hundreds or thousands of phones, split them up into groups no larger than what maybe 2 quad-span T1 (or E1) cards can handle (4*24= 96), times 2 is 192 lines per system, right? You'll have to figure out yourself how to split things up from there. It was suggested I obtain the latest, bleeding edge version of the kernel (nptl?), but I found that version wouldn't run X11 for some reason. So, I use the extent version of RH9, with all the neat RHN patches applied. The audio doesn't work at all, and I don't know why, but we'll leave that alone for now. Configuring channels and Order dependency. I have proven one thing: The order you declare your channels in the zaptel.conf file, and the order you load your modules MATTERS. Do it in the wrong order, and you will have problems. The instructions don't mention much if anything about this little detail. With this in zaptel.conf: fxsks=1,2 fxoks=3-6 I find that I should issue the commands in this sequence: modprobe wcfxs modprobe wcfxo ztcfg -v -v -v Swapping the order of the probes for fxs fxo will issue error messages when you run asterisk. If you run into this sort of channel allocation problem, reverse the order of your modprobes, but reboot between attempts. And, if all else fails, permute the order of the boards as they are plugged into your PCI slots. Oh, and even if you get the commands in the right sequence, you will most likely get some error or warning messages from the modprobes and/or the ztcfg... If asterisk runs OK, then these can be ignored. The existing documentation is very verbose about another possible point of confusion-- FXS hardware is signalled via FXO protocols, and vice versa for FXO hardware. Notice above that the FXS card has 4 ports, and is signalled via FXO signalling. And, the cards from Digium are all Kewlstart. You'll have to really root around on the web site for a while before you are certain of this. murf signature.asc Description: This is a digitally signed message part
[Asterisk-Users] Mediatrix 1204
I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has
gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway
working with asterisk. The Idea is the following.
PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} --
Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic
system has FX0 ports)
At the Hotel they dial there local extension lets say 1234 then the 1204 directs them
to our Asterisk which then sends the call to the working IVR. I need to get this
working with the least amount of hardware expense!
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[Asterisk-Users] Mediatrix 1204
I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has
gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway
working with asterisk. The Idea is the following.
PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} --
Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic
system has FX0 ports)
At the Hotel they dial there local extension lets say 1234 then the 1204 directs them
to our Asterisk which then sends the call to the working IVR. I need to get this
working with the least amount of hardware expense!
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[EMAIL PROTECTED]
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Re: [Asterisk-Users] Error in app_voicemail2.so after CVS update
On Wednesday 05 November 2003 12:03, Dan wrote: [app_voicemail2.so]WARNING[1074412256]: File loader.c, Line 232 (ast_load_resource): /usr/lib/asterisk/modules/app_voicemail2.so: undefined symbol: ast_localtime WARNING[1074412256]: File loader.c, Line 400 (load_modules): Loading module app_voicemail2.so failed! I don't know what you're doing wrong, because I just checked out CVS and tried this, and it works fine. ast_localtime() is located in the stdtime subdirectory of asterisk. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skinny (SCCP) help
I just got mine working. All I did was create a skinny.conf and point the phone to the asterisk server for tftp. the phone then boots and says useing TFTP as CM and works. I have no SEP.cnf's on my tftp server. my skinny.conf is [general] dateFormat = M-D-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 [will] device=SEP000750834016 context=default callerid=William carlson linelabel= mailbox= line = - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 1:24 PM Subject: [Asterisk-Users] Skinny (SCCP) help I have a cisco 7910 phone, I'm trying to get it to connect to asterisk, But it seems like it needs either a SEPDefault.cnf file or a SEPMACADDR.cnf file to Continue, I created empty ones but it's still sitting there saying opening Does anyone have examples of the SEPDefault.cnf file? Kevin, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
This company seems to think pros outweigh the cons for Asterisk: www.voicepulse.com /. reported today that VoicePulse uses a variation of Asterisk to run their Broadband Phone Service. http://slashdot.org/article.pl?sid=03/11/05/1319251mode=threadtid=126 Steven Critchfield wrote: On Wed, 2003-11-05 at 09:36, WipeOut wrote: Gavin Hamill wrote: It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' Can anyone think of any others? No built in high availability or clustering options making it as reliable as the harware, OS and apps.. Last time I looked it up PC systems combined hardware components average reliability was about 96% uptime(This was a while back so the percentage may not be accurate).. This is a problem for telecom's system whos uptime is usually measured in years and not a percentage of 1 year.. No flames please, I realise that there are issues involved with the PSTN lines, channel banks and some other things in a clustered senario.. I think the number you cited needs qualification to be accurate. Because if it where accurate as it stands, I'm due for major downtime in my rack as I have several systems approaching 2 years uptime without a single hardware failure. These machines also where not new when they where sent to the colo facility. In fact they all had been running for about a year before hand. And as a question of the 5 9's reported on telco hardware, As far as I know, that is for total system failure. The fact that they could loose trunks, or even a portion of a neighbor hood doesn't count against their downtime. If it did, I could point to a couple of telcos in this area that would have problems meeting those requirements. --- to back up my claim about uptime, my webserver is showing 136 days uptime, this is after a 497 day wrap around of the uptime counter. This machine is a Dell pe2450 the mail server is a home built 700 celeron showing the same 136 day uptime after the 497 day uptime wrap around. Due to a hacker, our clients machine is showing 105 days uptime post 497 day uptime wrap around. Again home built machine. One of our fileservers is showing 133 days uptime post uptime wrap around. This is due to a screw up at the keyboard just 3 days after installing it in the colo. Also a home built machine. Our VPN machine is just getting up to 354 days uptime. This is a super micro we purchased and put into service shortly there after. Our database server just went through a hardware and software upgrade that caused it's reboot, now at 185 days uptime. Same hardware as the above listed webserver. The 2 machines in my rack without impressive uptimes are a NT machine and my phone gateway that just had a kernel update. This should probe that good power supply to the machine will help make hardware run well for a long time. Why do you think the telco equipment runs on 48volts? They are pulling from the batteries 100% of the time. This makes a smooth even power flow. Machines in my office are subjected to poorer quality power and tweaking so they don't tend to make it to the 200 day uptime mark very often. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] archives gsm of asterisk ???
Hello. I have a problem I want to pass the archives of the voicemail of the a Spanish They can say to me that software I can use to create the archives gsm .. J.R
RE: [Asterisk-Users] g.729 codec registration
It is in fact G729A User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format 10 00070ea6-2f 00101/00103 0ms ms G729A 1 active SIP channel(s) Thanks, Brian On Wed, 5 Nov 2003, Thomas Haeger wrote: Hi i'am again... i have tesed if my * (where the purch. g729 is installed) take calls from a gateway with g.729A codec. The calling mechanism works but there is no voice only bad noises . I'am a little bit confused. On the digium site i bought a g729 codec (without any indication of an a or a b). I thought this codec could take calls with g729.a codec but this seems not to be so. If my fiction is right, how can i take calls with g.729.a codec ? Thanks, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Mittwoch, 5. November 2003 12:15 An: Asterisk User Betreff: [Asterisk-Users] g.729 codec registration Hi all, i have purchased the g.729 codec from digium. The registration was successful. (with the old binary) But there're a few questions: -should not the codec listed in the codec list when i enter show codecs ? -the codec is named with g729b but if i enter show codecs there is a codec g729a listed also the g729b is not installed. what is the difference between g729a built in * and the puchased g729b codec? Thanks for help. Regards, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
