RE: [Asterisk-Users] Bill seconds
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, 19 June 2005 2:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bill seconds next month and tell hoe much cell phone costs you. The point I was making is that the charges are NOT on _My_ cell phone bill, Why is it that if you pay 10 times as much to call a cell phone you consider it NOT part of your cell phone bill? Who says I do? Where did you pull that 10 times stuff? I don't have to pay anything more to call a Cell phone that I do to call a land line. In fact for the 5 mobiles that I own, (my family members) the calls between them and my land lines are free. Again, as the originator of the call I get to choose the amount I spend. Don't you see how they succeeded in making you believe that your cell phone is cheaper? I told you that none Amercians might not understand this. :) Yeah, I see how _some_ americans don't get it. when I don't originate the call, however in .us if you get called, you pay, that can easily cost you a heap of money that you can only control by switching the phone off, and where is the point in that? Really?? cost you a heap of money? only by swithcing the phone off? what ever happened to not picking up? Ok, there is that, so long as you take time to determine whether you recognise the number etc It does however make rec'ving calls on the Cell phone much less attractive. what about unlimited nights and weekends completely free that most providers give you here. What about the fact that even when you do pay for the incoming it costs around $.05 a minute? How about just not having to pay for incoming calls at all, that sounds much better. It makes being in touch easier and cheaper. I think I said enough. chuckle how does one respond to that? So if I rec'v 500 calls a week on my cell phone, it still costs me nothing. Wrong, because your provider succeeded in convincing your freind to make the same calculation, so when you have to call your friend you then pay 10 times as much than to a regular phone. Pure and unadulterated crapola, did you know that when people pluck numbers out of the air like that it belittles their entire point? And in some cases if I have the Cell and the Landline from the same telco (in .au), calls between them are free too, regardless of where I happen to be in australia at the time. So this we will take out of the argument since most American providers don't charge in network either. They do for out of zone calls, however with the telco I am using and the account arrangements I have, it doesn't matter where the cell phone is, even 4000km away is still a free call to my home land line. Oh, and cucumber seem to be doing you no favours either I can place a call to the US using my Cell phone for 1-2c/minute, shrug Caviat Emptor? Actualy you are right about this one, didn't realize they changed the rates to au, it used to be $.039 a minute. Thanks for pointing this out. In any case I know that Australia has now very good rates to call UK and the states, but that is only as far as LD goes. I have VoIP for calls to the .us and .uk I also can route my call via my home * box and then over VoIP to many other places to make the calls *free* so with a call to .us for instance, I can use my cell to call one of my home land lines *free* and then via * connect to the us using one of the IP Telcos *1c/min* , or to my office in Houston to the * box there *free* Further: In the .US there is a groundswell of people that are angry with telemarketers calling them on their cell phones, Why is this? A: because the cost of the call is shifted to the called party, just like spam. The .au model of caller pays has pretty much ensured that telemarketers wont be a problem on _my_ cell phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on vpbx exited on signal 11. Might want to take a peek.
Asterisk on vpbx exited on signal 11. Might want to take a peek. Got above email, ... Where and what should I check? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] chan_capi-cm-0.5 release announcement
On Sun, 19 Jun 2005, Stefan Gofferje wrote: Armin Schindler schrieb: I would like to announce the first release of the chan_capi channel driver on sourceforge.net The package is available for download with name chan_capi-cm-0.5 and is the current CVS HEAD. It is derived from the chan_capi-0.4.0PRE1 of kapejod. The main changes are: - complete rework - fix race-conditions - fix call state handling - rework of debug/verbose messages - added capiFax feature (provided by Frank Sautter) - auto-config (compile and work with Asterisk CVS-HEAD and older versions) - use with ELinOS cross-toolbox and project handling Very nice...! Does your CAPI channel handles busy/congestion correctly? chan_capi-0.3.5 does neither detect and report busy / unavailable messages from the PSTN nor does it handle busy() / congestion() commands and report the proper state to an external caller... Not yet, but this is actually the next thing on my list. I will soon release a fixup version. I found two more bugs, a memory leak and voice-buffer-corruption on SMP systems. Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is actually somebody doing it: load balancing 20 asterisk servers
load balancing 20 asterisk servers http://lists.digium.com/pipermail/asterisk-users/2005-February/087301.html I read the thread, but I am on the point zero again. Is anybody actually doing it? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Now it is working
Hi All, I am very happy to say that after solving a lot of problems I got my Asterisk box talking (with a Digium's TDM20B card) last 48hrs I was sending message to many people asking help. I found some solutions myself but the base of those solutions given by the generous people in the list. I appriciate all of them once again. Now, I have minor problems like jitter and fine-tuning, I am going to start that part little by little, Please give me your advices, ideas, experience on this process of fine tuning.The dream I was looking in the last 5-6 months has become real. Thanks to everybody who help me directly and indirectly (I steel things from the list). and I would like to state that if I see any question in the list that I could give an idea on it. I would do it. Thank you all Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] chan_capi-cm-0.5 release announcement
On Sun, 19 Jun 2005, Stefan Gofferje wrote: Armin Schindler schrieb: On Sun, 19 Jun 2005, Stefan Gofferje wrote: Armin Schindler schrieb: I would like to announce the first release of the chan_capi channel driver on sourceforge.net The package is available for download with name chan_capi-cm-0.5 and is the current CVS HEAD. It is derived from the chan_capi-0.4.0PRE1 of kapejod. The main changes are: - complete rework - fix race-conditions - fix call state handling - rework of debug/verbose messages - added capiFax feature (provided by Frank Sautter) - auto-config (compile and work with Asterisk CVS-HEAD and older versions) - use with ELinOS cross-toolbox and project handling Very nice...! Does your CAPI channel handles busy/congestion correctly? chan_capi-0.3.5 does neither detect and report busy / unavailable messages from the PSTN nor does it handle busy() / congestion() commands and report the proper state to an external caller... Not yet, but this is actually the next thing on my list. I will soon release a fixup version. I found two more bugs, a memory leak and voice-buffer-corruption on SMP systems. Cool! Is there a mailinglist or a forum for chan_capi-cm? No, if necessary we can activate a mailinglist via sourceforge. But currently I activated the bug-tracker only. Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER and Asterisk question
Dear All, I just solved my problem, you can make asterisk itself make phones peer with each other if they support the canreinvite option, so step number one was to insert this option in the sip.conf configration in the phones part: [xxx] canreinvite=yes Now the calls will be direct between the two IP Phones without having asterisk in the middle which will save bandwidth on the wan link. As for SER when you perform it after this step it shoild work fine with you. Thx MAG Mohamed A. Gombolaty wrote: Dear Yair, Actually what happens is that from SER debug I can see the call is looping between Asterisk and SER. but adding a number makes no loops. Thx MAG Yair Hakak wrote: yes, there is. run everything through asterisk, no matter how long the extensions are. for example, 666 calls 999 goes to asterisk, sees a dial sip:[EMAIL PROTECTED], goes back to SER. bounces back to ser. If everything is working well asterisk will set up the call and get out of the way. I don't see why you need to prepend digits in order to make this work, if i'm missing something let me know. -yair On 6/16/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote: Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and which has a uri rule that says forward 4 digit starting with 1 to the asterisk sip port the asterisk extensions.conf has an entry for 1999 and dials [EMAIL PROTECTED], if not answered voicemail runs and so on. ain't there a way to make 666 directly call 999 without using 1999. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2
On Friday 17 Jun 2005 18:05, Manuel Casal wrote: Marco Parmeggiani escribi: Manuel Casal ha scritto: I made the make menuconfig and make dep in the kernel sources. i do not remember well how i solved that problem but i'm sure that make dep will issue you a warning and stop. run make to start the kernel build process and then stop it after few seconds. it will create the necessary symlinks in the kernel tree. maybe there's a more elegant solution but this should work. [... Yet another f*ck*ng signature not deleted ...] [...] SUBDIRS=/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 modules make[1]: Entering directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp' make: *** [linux26] Error 2 linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 # Now what?:( Have a read at /usr/src/linux*/README.SUSE ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't switch span to E1-mode
On Friday 17 Jun 2005 17:15, Yousef Herzallah wrote: Hi, This error I got it just when I gonfigure zaptel support isdneuro 31 channels. But if I configure zaptel to support T1 and just 24 channels I have no problem. # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = it span=1,1,0,ccs,hdb3,crc4 bchan=1-23 dchan=16 bchan=17-31 First, the B channels run from 1-15 17-31, secondly , flip the jumpers on the card. [...] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] stale nonce received
What does it mean? Jun 19 20:18:39 NOTICE[17484]: chan_sip.c:5472 check_auth: stale nonce received from 'sip:[EMAIL PROTECTED]' bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Console ALSA Sound
Sean Kennedy wrote: Heh, try asking about line appearances and the hint priority. People clam right up. Or ask about receptionist phones that show all your line statuses. You can practically hear the crickets. :) A quick search through the ML folder told me you asked about that on May 31. I've set up line status on Snom phones a few days ago, using HEAD from May 29 and bristuff, and I didn't have any luck until I realised that the support was broken in pbx.c: maybe that was your problem too. It has been alredy fixed in CVS :) I haven't tried with stable. -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: rxfax problem - libspandsp issue?
They keep breaking the FAX support in libtiff. 3.6.1 is broken, a ... Hi, thanks for the information about libtiff 3.6.1. I had to search a while in order to find the old libtiff 3.5.7, which now works fine in my asterisk installation. For those being in the need of libtiff 3.5.7: Pay attention to my subsequent email! Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Libtiff 3.5.7 - recommended version for spandsp
Hi, package tiff-v3.5.7 contains the currently recommended version of libtiff in order to run spandsp (fax support for asterisk). Imho tiff-v3.5.7 is not very easy to find in the internet, and maybe will almost disappear, because it is an old version, I put it on our little asterisk download page. Maybe it'll help someone. It works fine together with the other asterisk stuff (around version 1.0.7) located in that directory: http://planinternet.net/download/voip/asterisk Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bluetooth audio and asterisk
Has anyone successfully used a standard bluetooth enabled system to connect to a standard bluetooth enabled mobile phone (not the bluetooth to FXS converters) to create an audio path for phone calls with asterisk, if so is there a writeup on what was done so that others can replicate this. What I am thinking is that via alsa/oss/whatever you should be able to use the bluetooth audio channel as a speaker and microphone to talk to a mobile. The catch is of course sending dialing information via bluetooth from within asterisk to cause this to properly occur. SMS could also be sent/received via bluetooth and converted to SIP IM for example thus giving more devices accessability to asterisk 'out of the box'. And given cost, many bluetooth dongles can be purchased for $30 USD or less, this would create a lower entry price for people to use this technology. Cellsocket.com for example is a phone-FXS adapter, they are $100. Someone this last week mentioned a bluetooth system that is similar, but afaik both devices do not allow for SMS to be trafficed via asterisk, and both cost more than a bluetooth dongle. I know that linux does support bluetooth via qualcomms BT API, and I would imagine that other systems also support this. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_curl.so: Can't locate module sound-service-0-3
Hi, What does an't locate module sound-service-0-3 mean in the context below? I don't use curl as far as I know. Should I noload it in modules.conf? [app_curl.so]Jun 19 08:41:25 b2 kernel: Registered tone zone 0 (United States / North America) Jun 19 08:41:34 b2 kernel: Intel 810 + AC97 Audio, version 1.01, 02:25:38 Dec 24 2004 Jun 19 08:41:34 b2 kernel: PCI: Found IRQ 10 for device 00:1f.5 Jun 19 08:41:34 b2 kernel: PCI: Sharing IRQ 10 with 00:1f.3 Jun 19 08:41:34 b2 kernel: i810: Intel ICH2 found at IO 0xdc40 and 0xd800, MEM 0x and 0x, IRQ 10 Jun 19 08:41:34 b2 kernel: i810_audio: Audio Controller supports 6 channels. Jun 19 08:41:34 b2 kernel: i810_audio: Defaulting to base 2 channel mode. Jun 19 08:41:34 b2 kernel: i810_audio: Resetting connection 0 Jun 19 08:41:34 b2 kernel: ac97_codec: AC97 Audio codec, id: ADS96 (Analog Devices AD1885) Jun 19 08:41:34 b2 kernel: i810_audio: AC'97 codec 0 Unable to map surround DAC's (or DAC's not present), total channels = 2 Jun 19 08:41:34 b2 kernel: i810_audio: setting clocking to 41349 Jun 19 08:41:34 b2 kernel: i810_audio: ftsodell is now a deprecated option. Jun 19 08:41:34 b2 modprobe: modprobe: Can't locate module sound-service-0-3 = (Load external URL) == Registered application 'Curl' -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] missing mysql cdr records
I installed asterisk on two machine to load balance and to fail over. One is local and one is on a remote location. The record the cdr I use the mysql module and odbc. So I update a local (on each asterisk box) mysql db using mysql module and also I update a centralized mysql db using ODBC. I tought in this way I couldn't loose any record. But this is not true. Even with 2 updating process I loose around 1%-2% of the records. I'm still in a testing enviroment so it is not a problembut asap i will be in production. I have a cdr report from my upstream and also I log the sip INVITE ACK and BYE messages so I am pretty sure that the reports from my upstream is accurate and my mysql cdr misses some record. I find that also the local db (on the asterisk box, very low load) misses some record. I upped the max_connection to 2000 on mysql but didn't help. What can I check to find out what cause my problem? Thanks Rosario ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I don't get a queue_log with my version of asterisk (0.7.1).
Dear All, I would like to add this feature to my version of asterisk. Is there a patch I can apply to get this function? Does anyone have any instructions for this? Thanks Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] bluetooth audio and asterisk
Do a search on voip-info, Also there is an Australian company that build Bluetooth base stations that connect to a pabx (quasi dect cell based system) but big bucks (though does prove it is technically feasible. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Sunday, 19 June 2005 8:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bluetooth audio and asterisk Has anyone successfully used a standard bluetooth enabled system to connect to a standard bluetooth enabled mobile phone (not the bluetooth to FXS converters) to create an audio path for phone calls with asterisk, if so is there a writeup on what was done so that others can replicate this. What I am thinking is that via alsa/oss/whatever you should be able to use the bluetooth audio channel as a speaker and microphone to talk to a mobile. The catch is of course sending dialing information via bluetooth from within asterisk to cause this to properly occur. SMS could also be sent/received via bluetooth and converted to SIP IM for example thus giving more devices accessability to asterisk 'out of the box'. And given cost, many bluetooth dongles can be purchased for $30 USD or less, this would create a lower entry price for people to use this technology. Cellsocket.com for example is a phone-FXS adapter, they are $100. Someone this last week mentioned a bluetooth system that is similar, but afaik both devices do not allow for SMS to be trafficed via asterisk, and both cost more than a bluetooth dongle. I know that linux does support bluetooth via qualcomms BT API, and I would imagine that other systems also support this. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bluetooth audio and asterisk
Hello, Has anyone successfully used a standard bluetooth enabled system to connect to a standard bluetooth enabled mobile phone (not the bluetooth to FXS converters) to create an audio path for phone calls with asterisk, if so is there a writeup on what was done so that others can replicate this. I want to do the same thing. I want to buy an bluetooth capable mobile phone without a dongle. I want asterisk to predict it's a headset, so it can route audio. This way, I could accept calls from mobile network for very cheap. And also do the opossite route (call to cell-phone network). Check this: http://www.voip-info.org/tiki-index.php?page=Asterisk+Bluetooth+channels If you'll have any progress in this area, please let me know. It should also be able to send SMS, I have previously done this successfully, but without Asterisk using smstools. For sending sms, using a web gateway would be probably cheaper, but it should be good for receiving. Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and fayn.cz
Hello, I would like to use Asterisk with fayn.cz service. They should be using a standard H.323 protocol, but I have no more information about this. They provide a softphone and/or rebranded H.323 telephone, but I don't know any H.323 settings nor if the firmware in the phone is modified. Has anyone tried this successfully? They provide a Prague telephone number reachable from classic telephone network in their free monthly plan, which would be of great use for me, since I have many friends in Czech Republic and it would be very cheap to call to a local prague number instead of international call. Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Presence and IM?
Hello, We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se who's online and available and who's not. Surely, there's the manager interface, but unless you'd want to install extra software on each client computer, this is not a good option. Then there's the presence / IM function in SIP. Since we're only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? I have the same problem and am seeking for few weeks for a suitable solution... If you'll figure out something, please let me know. We use Polycom IP500s which when used with a 'hint' in extensions.conf, can show presence via the 'buddy list.' could you post a snippet? Does this hint work as a presence agent and sending notifies? Does IM work with asterisk? I would really like to support presence in Asterisk with Eyebeam as a client. SIP Express Router has this ability, but it's not a good choice either. Maybe it would be possible to port this feature from SER? Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi IP Phones
Hello, I know there are wifi sip phones out there but I have a question, are any of these phones anti explosive? By that I mean, there are certain regulations about phones or cel phones that are not recommended to operate in environments like gas stations due to sparks and the chance of ingiting gas fumes. BTW: it would probably be even cheaper to use a normal analogue wireless phone and connect it's base station to FXO port. You can get cordless analogue telephones for cheap these days and I believe when you buy a card with FXO to connect the base station to them, it would be even cheaper than buying a Wifi phone... Have you considered this? Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi IP Phones
On Sun, 2005-06-19 at 19:22 +0200, Juraj Bednar wrote: Hello, I know there are wifi sip phones out there but I have a question, are any of these phones anti explosive? By that I mean, there are certain regulations about phones or cel phones that are not recommended to operate in environments like gas stations due to sparks and the chance of ingiting gas fumes. BTW: it would probably be even cheaper to use a normal analogue wireless phone and connect it's base station to FXO port. You can get cordless analogue telephones for cheap these days and I believe when you buy a card with FXO to connect the base station to them, it would be even cheaper than buying a Wifi phone... Have you considered this? A little bulkier to take with you to a coffee shop or other public wifi facility :P -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bill seconds
Why is it that if you pay 10 times as much to call a cell phone you consider it NOT part of your cell phone bill? Who says I do? Where did you pull that 10 times stuff? I don't have to pay anything more to call a Cell phone that I do to call a land line. In fact OK, here it looks like you either misuderstood somthing or you are a liar. I already included the link where it showed it costs more to call a cell phone. As for the the 10 times figure I made a mistake (since I was still under the impression that it costs only $.039 to call australia landline) and make it 4+ times as much (7 cents to landline and 30 to cell, that makes; 30/7=4+2/7 times as much as to a landline). for the 5 mobiles that I own, (my family members) the calls between them and my land lines are free. You already mentioned that (see below) that is NOT the argument. Again, as the originator of the call I get to choose the amount I spend. Don't you see how they succeeded in making you believe that your cell phone is cheaper? I told you that none Amercians might not understand this. :) Yeah, I see how _some_ americans don't get it. when I don't originate the call, however in .us if you get called, you pay, that can easily cost you a heap of money that you can only control by switching the phone off, and where is the point in that? Really?? cost you a heap of money? only by swithcing the phone off? what ever happened to not picking up? Ok, there is that, so long as you take time to determine whether you recognise the number etc It does however make rec'ving calls on the Cell phone much less attractive. I totaly agree that it makes it unattractive, but in no way does't it make the person calling me hesitate, so I can realy keep in touch. what about unlimited nights and weekends completely free that most providers give you here. What about the fact that even when you do pay for the incoming it costs around $.05 a minute? How about just not having to pay for incoming calls at all, that sounds much better. It makes being in touch easier and cheaper. Maybe, it makes it easier for the receiver but not for the one making the call. So this part is again debateable, and not what the argument is about. But if you add up the cents and dollars it is cheaper to use cell phones in the states - where incoming costs sometimes as little as making a LD domestic call for the owner of the cell phone - than it is in Australia, or all the other countries that they charge as much as 4+ times to call the cell network. I think I said enough. chuckle how does one respond to that? So if I rec'v 500 calls a week on my cell phone, it still costs me nothing. Wrong, because your provider succeeded in convincing your freind to make the same calculation, so when you have to call your friend you then pay 10 times as much than to a regular phone. Pure and unadulterated crapola, did you know that when people pluck numbers out of the air like that it belittles their entire point? Can you explain why you can't argue this in english? or is it that you see that I am right? Now the only thing that I made a mistake about is the 10 times it should be 4+ times. And in some cases if I have the Cell and the Landline from the same telco (in .au), calls between them are free too, regardless of where I happen to be in australia at the time. So this we will take out of the argument since most American providers don't charge in network either. They do for out of zone calls, however with the telco I am using and the account arrangements I have, it doesn't matter where the cell phone is, even 4000km away is still a free call to my home land line. Really? I have a cell phone here in the states since January 1998, I have had cell phones with: Verizon, SprintPCS, ATT, Cingular, and Nextel. None of them ever had so called out of zones, as long as I was anywhere on their network (CA to NY, to FL to WA, and all of the lower 48) had the same rate. In my family we currently have more than 10 cell phones, none pay any extra based on where they are. Here show me how many plans have what you describe: http://www.sprintpcs.com/ http://www.verizonwireless.com/b2c/index.jsp http://www.nextel.com/ http://www.cingular.com/indexc http://www.t-mobile.com/ Oh, and cucumber seem to be doing you no favours either I can place a call to the US using my Cell phone for 1-2c/minute, shrug Caviat Emptor? Actualy you are right about this one, didn't realize they changed the rates to au, it used to be $.039 a minute. Thanks for pointing this out. In any case I know that Australia has now very good rates to call UK and the states, but that is only as far as LD goes. I have VoIP for calls to the .us and .uk I also can route my call via my home * box and then over VoIP to many other places to make the calls *free* so with a call to .us for
[Asterisk-Users] chan_capi-cm-0.5.1 fixup release
Hi all, on sourceforge.net I added the fixup release 0.5.1 of chan_capi-cm driver. The changes from 0.5 to 0.5.1 are: - fixed a memory leak (in ast_smoother usage) - fixed voice buffer corruption on SMP systems (each channel now has its own buffers) - removed unused variables Have fun Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *67 with Sipura 3000
How can I dial *67 on a Sipura 3000 if I dial from a SIP phone
connected on an asterisk server. I always get a message saying that
authentication failed for INVITE for [EMAIL PROTECTED] If I dial a
number without doing *67 it's working fine...
sip 221 being the extension of my Cisco phone and 192.168.1.6 being
the IP of my asterisk server...
I have my outgoing context configure like this :
[outgoing]
ignorepat = 9
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9.,4,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9.,5,Playback(invalid)
exten = _9.,6,Hangup
When I do 9*67 and the number it take a while and then it will play
the invalid sound file and then hangup.
I even tried adding a second outgoing with this but it doesn't make a
difference :
exten = _9*67.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9*67.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9*67.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9*67.,4,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9*67.,5,Playback(invalid)
exten = _9*67.,6,Hangup
I figured that's it is a function of the Sipura 3000 but can I
disable it and make it seen as a number? Since Bell understand *67 I
don't need the Sipura 3000 to do something special with it...
Thanks
Martin
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[Asterisk-Users] MySQL to static .conf
Hello, I'm running asterisk 1.0.7 on a kernel 2.4 linux box. I have my SIP users database on MySQL, also I'm running asterisk from static .conf files. What I want to get done is that after a pre defined period of time, my SIP users table must be converted to a .conf file. Perhaps with a script and cron entry. Please shed me some light on this. Thanks in advance. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel and Zapata Conf's
I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure how to show channels numbers for the FXS Ports. This was working properly before I upgraded to TDM400P (was X100P). But now the TE410P LED's are flashing RED Please Help Bart Here is my current Zaptel.conf: # Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 AMI/D4 span=1,0,0,d4,ami em=1-24 # Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 AMI/D4 RED span=2,0,0,d4,ami em=25-48 # Span 3: TE4/0/3 TE410P (PCI) Card 0 Span 3 AMI/D4 RED span=3,0,0,d4,ami em=49-72 # Span 4 TE4/0/4 TE410P (PCI) Card 0 Span 4 AMI/D4 RED span=4,0,0,d4,ami em=73-96 # Span 5: WCTDM/0 Wildcard TDM400P REV E/F Board 1 fxoks=97 fxoks=98 # channel 3, WCTDM, inactive. # channel 4, WCTDM, FXO fxsks=99 And Current Zapata.conf: ; Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 ; This T1 is attached to in-house CUST 3 System ; language=en rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 immediate=no busydetect=no busycount=15 callprogress=no ;relaxdtmf=yes ;callerid=asreceived faxdetect=incoming signalling =em_w group = 2 channel = 1-24 ; Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 ; This T1 is attached to inhouse CUST 10 System ; language=en rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 immediate=no busydetect=no busycount=15 callprogress=no ;relaxdtmf=yes ;callerid=asreceived faxdetect=incoming signalling =em_w group = 3 channel = 25-48 ; Span 3: TE4/0/3 TE410P (PCI) Card 0 Span 3 ; This T1 is attached to WorldCom Local 714 DID's language=en context=from-localt1 ; = rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 immediate=no busydetect=no busycount=15 callprogress=no ;relaxdtmf=yes ;callerid=asreceived faxdetect=incoming signalling =em_w group = 4 channel = 49-72 ; Span 4 TE4/0/4 TE410P (PCI) Card 0 Span 4 ; GBX inbound outbound T1 language=en context=from-tollfree ; = rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 immediate=no busydetect=no busycount=15 callprogress=no ;relaxdtmf=yes ;callerid=asreceived faxdetect=incoming signalling =em_w group = 5 channel = 73-96 ; Span 5: WCTDM/0 Wildcard TDM400P REV E/F Board 1 ; ; Note: this is an extension. Create a ZAP extension in AMP ; for Channel 1 ; signalling=fxo_ks context=from-internal group=1 channel = 97 ; ; Note: this is an extension. Create a ZAP extension in AMP ; for Channel 2 ; signalling=fxo_ks context=from-internal group=1 channel = 98 ; ; channel 3, WCTDM, inactive. ; ; Note: this is a trunk. Create a ZAP trunk in AMP ; for Channel 4 ; signalling=fxs_ks context=from-pstn group=0 channel = 99 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with astperl primitives say... in astcc
I just upgraded to the latest (as of a week ago) CVS and since them, I've had a problem with astcc. I've traced the problem as far as astcc calling any of the AGI say... routines (say_digits, say_number, etc.). As near as I can tell, the calls are executed, but control never returns to the astcc code that made the call, and as a result, the channel simply hangs (i.e., nothing else happens) and astcc never returns to the dialplan. Has anyone else experienced this or anything like it? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chanisavail...not workin with SIP and IAX
all I cannot get ChanIsAvail to work with sip or iax on v1.0.3. It does work fine on a zap channel. I am trying with Sipuras and PAP2s. It appears I am not the only one having this problem. Has anyone gotten it to work? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NAT/Proxy advise
Hi all, Am looking for everyones advise/recommendations. I have am setting up a network of both office and home based workers. The office workers will be on the same network as the Asterisk box so no NAT hassles there. However, the home workers are on their own DSL connections so I imagine that most will be behind NAT. I had hoped to use Sipura SPA-2000's but I presume I would need either STUN or a proxy to make these work if behind NAT. Please could I ask for any recommendations as to which approach would be best and what software/hardware people recommend. Thanks for any help and advise, Neil ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tos problem
Hello people, It seems that my * does not react to tos=whatever field in iax.conf. I am using latest CVS HEAD code. Can anybody help me with this issue? ps: if i go to chan_iax2.c and modify the initial definition of tos variable, it works fine marking packets with the value specified there: static int tos=16; if i put random text in iax.conf's tos=, chan_iax2 refuses to load because of incorrect value (so, it reads the value from the file, but is unable to set it properly) thanks, Calin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tos problem
Calin Serbanescu wrote: static int tos=16; I think it is tos=0x16 Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [SOLVED IAX with shaw cable not going through
It seems to me spoofing MAC address is causing the problem. I've connected the original firewall that I tested (without spoof MAC address assigned to firewall) and every connection is working FWD, VoipJet. It seems it me that new Shaw Cable - Motorola SURFboard SB5100 is a piece or crap. The solution was to unplug the Sipura from the power for few seconds and plug it back to the wall. One of those silly things. The Sipura 3000 unit was working locally just fine, calls could go through IN/OUT to FXO but it wouldn't connect through firewall. Unplugging it it from the power supply solved the problem. Don't ask me why! -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bill seconds
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 June 2005 3:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bill seconds Why is it that if you pay 10 times as much to call a cell phone you consider it NOT part of your cell phone bill? Who says I do? Where did you pull that 10 times stuff? I don't have to pay anything more to call a Cell phone that I do to call a land line. In fact OK, here it looks like you either misuderstood somthing or you are a liar. Oooo, a diplomat! :) I showed you that your link to a mob called cucumber was not helpful to you or anyone else. Their pricing is fure fiction as far as .au telco pricing is concerned. I already included the link where it showed it costs more to call a cell phone. As for the the 10 times figure I made a mistake (since I was still under the impression that it costs only $.039 to call australia landline) and make it 4+ times as much (7 cents to landline and 30 to cell, that makes; 30/7=4+2/7 times as much as to a landline). That's what happens when you pull figures out of the air. chuckle for the 5 mobiles that I own, (my family members) the calls between them and my land lines are free. You already mentioned that (see below) that is NOT the argument. Why not? Again, as the originator of the call I get to choose the amount I spend. Don't you see how they succeeded in making you believe that your cell phone is cheaper? I told you that none Amercians might not understand this. :) Yeah, I see how _some_ americans don't get it. when I don't originate the call, however in .us if you get called, you pay, that can easily cost you a heap of money that you can only control by switching the phone off, and where is the point in that? Really?? cost you a heap of money? only by swithcing the phone off? what ever happened to not picking up? Ok, there is that, so long as you take time to determine whether you recognise the number etc It does however make rec'ving calls on the Cell phone much less attractive. I totaly agree that it makes it unattractive, but in no way does't it make the person calling me hesitate, so I can realy keep in touch. And so your spending level is dictated to you buy people that want to call you, at the whim of another (so to speak) what about unlimited nights and weekends completely free that most providers give you here. What about the fact that even when you do pay for the incoming it costs around $.05 a minute? How about just not having to pay for incoming calls at all, that sounds much better. It makes being in touch easier and cheaper. Maybe, it makes it easier for the receiver but not for the one making the call. And it is the one that _chooses_ to make the call that make the decision to spend the money. Who's money should they be able to choose to spend? Quite frankly someone else being able to spend my money at their whim scares the willies out of me. So this part is again debateable, and not what the argument is about. But if you add up the cents and dollars it is cheaper to use cell phones in the states - where incoming costs sometimes as little as making a LD domestic call for the owner of the cell phone - than it is in Australia, or all the other countries that they charge as much as 4+ times to call the cell network. So the caller is more likely to (a) not waste my time, (b) not waste my money, (c) Get on with what they wanted to tell me, etc. I think I said enough. chuckle how does one respond to that? So if I rec'v 500 calls a week on my cell phone, it still costs me nothing. Wrong, because your provider succeeded in convincing your freind to make the same calculation, so when you have to call your friend you then pay 10 times as much than to a regular phone. Pure and unadulterated crapola, did you know that when people pluck numbers out of the air like that it belittles their entire point? Can you explain why you can't argue this in english? or is it that you see that I am right? Now the only thing that I made a mistake about is the 10 times it should be 4+ times. And in some cases if I have the Cell and the Landline from the same telco (in .au), calls between them are free too, regardless of where I happen to be in australia at the time. So this we will take out of the argument since most American providers don't charge in network either. They do for out of zone calls, however with the telco I am using and the account arrangements I have, it doesn't matter where the cell phone is, even 4000km away is still a free call to my home land line. Really? I have a cell phone here in the states
Re: [Asterisk-Users] Unable to make outbound calls
i hv set the verbose level to 4 and this is the output. - Accepting AUTHENTICATED call from 192.168.0.64, requested format = 2, act ual format = 2 -- Executing Macro(IAX2/[EMAIL PROTECTED]/4, dialout-trunk|1|7857303|) in new stac k -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/4, 1?3:2)) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(IAX2/[EMAIL PROTECTED]/4, record-enable|201|OUT) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/4, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/4, 1?5:8) in new stack -- Goto (macro-record-enable,s,5) -- Executing DBget(IAX2/[EMAIL PROTECTED]/4, RecEnable=RECORD-OUT/201) in new stac k -- DBget: varname=RecEnable, family=RECORD-OUT, key=201 -- DBget: Value not found in database. -- Executing SetVar(IAX2/[EMAIL PROTECTED]/4, CALLFILENAME=OUT201-20050619-112719-1 119205639.21) in new stack -- Executing Goto(IAX2/[EMAIL PROTECTED]/4, s|14) in new stack -- Goto (macro-record-enable,s,14) -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/4, 0?15:99) in new stack -- Goto (macro-record-enable,s,99) -- Executing NoOp(IAX2/[EMAIL PROTECTED]/4, NO RECORDING NEEDED) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/4, 1?7) in new stack -- Goto (macro-dialout-trunk,s,7) -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/4, 0?9) in new stack -- Executing SetCallerID(IAX2/[EMAIL PROTECTED]/4, 14253926763) in new stack -- Executing SetGroup(IAX2/[EMAIL PROTECTED]/4, OUT_1) in new stack -- Executing CheckGroup(IAX2/[EMAIL PROTECTED]/4, 1) in new stack -- Executing SetVar(IAX2/[EMAIL PROTECTED]/4, DIAL_NUMBER=7857303) in new stack -- Executing SetVar(IAX2/[EMAIL PROTECTED]/4, DIAL_TRUNK=1) in new stack -- Executing AGI(IAX2/[EMAIL PROTECTED]/4, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(IAX2/[EMAIL PROTECTED]/4, OUTNUM=7857303) in new stack -- Executing Cut(IAX2/[EMAIL PROTECTED]/4, custom=OUT_1|:|1) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/4, 0?19) in new stack -- Executing Dial(IAX2/[EMAIL PROTECTED]/4, ZAP/g1/7857303) in new stack == Everyone is busy/congested at this time -- Executing Goto(IAX2/[EMAIL PROTECTED]/4, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(IAX2/[EMAIL PROTECTED]/4, Dial failed due to CHANUNAVAIL) in new stack -- Executing Macro(IAX2/[EMAIL PROTECTED]/4, outisbusy) in new stack -- Executing Playback(IAX2/[EMAIL PROTECTED]/4, allison7/all-circuits-busy-now) in new stack -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback(IAX2/[EMAIL PROTECTED]/4, allison7/pls-try-call-later) in ne w stack -- Playing 'allison7/pls-try-call-later' (language 'en') -- Executing Macro(IAX2/[EMAIL PROTECTED]/4, hangupcall) in new stack -- Executing ResetCDR(IAX2/[EMAIL PROTECTED]/4, w) in new stack -- Executing NoCDR(IAX2/[EMAIL PROTECTED]/4, ) in new stack -- Executing Wait(IAX2/[EMAIL PROTECTED]/4, 5) in new stack -- Executing Hangup(IAX2/[EMAIL PROTECTED]/4, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'IAX2/[EMAIL PROTECTED]/4 ' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'IAX2/[EMAIL PROTECTED]/4' Some more info asterisk1*CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-pstn en 1from-pstn en zap show channel 1 asterisk1*CLI zap show channel 1 Channel: 1CLI File Descriptor: 15 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID string: Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Actual Hookstate: Offhook asterisk1*CLIshow modules chan_zap.so Zapata Telephony w/PRI 0 Appreciate your help. -B --- Balaji NJL [EMAIL PROTECTED] wrote: Hi All, I am a new bee to *. I just installed [EMAIL PROTECTED] on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message All routes are busy. In the asterisk
Re: [Asterisk-Users] Want to test my * behind firewall
Can someone connect to my server and leave a message. i appreciate it. -B --- Balaji NJL [EMAIL PROTECTED] wrote: Can someone leave a message at x 200 on my * server. External IP two one six . nine . zero . three four Connect as x 202 password zxc123 using IAX2 thanks, -B __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Discover Yahoo! Use Yahoo! to plan a weekend, have fun online and more. Check it out! http://discover.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT/Proxy advise
Neil Bullock wrote: Hi all, Am looking for everyones advise/recommendations. I have am setting up a network of both office and home based workers. The office workers will be on the same network as the Asterisk box so no NAT hassles there. However, the home workers are on their own DSL connections so I imagine that most will be behind NAT. For this I'd recommend a VPN (just use the standard MS PPTP VPN as it's extremely easy for non-techies to setup or be talked through over the phone). Then you don't have to worry about NAT settings. You can also encrypt the tunnel, if security is an issue. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bill seconds
I showed you that your link to a mob called cucumber was not helpful to you or anyone else. Their pricing is fure fiction as far as .au telco pricing is concerned. Really pure fiction? Mob? let see: http://www.tel3advantage.com/rates.aspx?AgentNumber=036333CID=124 $.03 to regulare, and $.17 to mobile (more than 5 times as much) http://www.packet8.net/about/international.asp again $.03 to regular, and $.23 cents to mobile more than 7 times as much http://www.broadvoice.com/rateplans_international_li.html $.02 to regular, and $.18 to mobile 9 times as much http://www.voicepulse.com/plans/InternationalRates.aspx $.06 to regular and $.26 to mobile, that makes more than 4 times as much. anyhow to show you that cucumber is not the most expensive one: http://www22.verizon.com/ForYourHome/sas/sas_con_LongDescription.aspx $1.30 to australia here is one thats even better: http://www22.verizon.com/ForYourHome/sas/sas_basicinternationalcallingcardrates.aspx Here is another Verizon rate: http://www22.verizon.com/ForYourhome/voip/CallingRates.aspx Don't ask me why the difference, but I promise you they don't even know. I already included the link where it showed it costs more to call a cell phone. As for the the 10 times figure I made a mistake (since I was still under the impression that it costs only $.039 to call australia landline) and make it 4+ times as much (7 cents to landline and 30 to cell, that makes; 30/7=4+2/7 times as much as to a landline). That's what happens when you pull figures out of the air. chuckle Really out of the air? the interesting part here is that you know better than me that a huge chunk of your monthly phone bill (not your cell phone) goes towards phone calls made to mobile phones, which is something that in the states doesn't exist, and still you argue that it doesn't cost you, and you divert this argument about what some company charges to Australia. In an avarage month every American can tell you EXACTLY how much they are GOING to pay for their cellphone that month, and in most cases it is not a lot based on the minutes used. However in places like Australia that you pay for your cell phone when calling from your home phone, there is no way of telling how much it is costing you since it costs you sometimes as much as 9 times as much to call a cell phone. for the 5 mobiles that I own, (my family members) the calls between them and my land lines are free. You already mentioned that (see below) that is NOT the argument. Because basic math teaches us that 2 negatives cancel each other, and I told you that the same is available in the states, so this argument is negated with the exact same argument that I have, and that is that I don't have to pay to ANY customer that is in the same network that I am (currently SprintPCS) nor does he pay for the incoming. So far all you have is only 5, and in the states I get about 30 Million phone numbers that I can call for free UNLIMITED (besides for nights and weekends that are completely free), so if you want this is another one for me. Again, as the originator of the call I get to choose the amount I spend. Don't you see how they succeeded in making you believe that your cell phone is cheaper? I told you that none Amercians might not understand this. :) Yeah, I see how _some_ americans don't get it. when I don't originate the call, however in .us if you get called, you pay, that can easily cost you a heap of money that you can only control by switching the phone off, and where is the point in that? Really?? cost you a heap of money? only by swithcing the phone off? what ever happened to not picking up? Ok, there is that, so long as you take time to determine whether you recognise the number etc It does however make rec'ving calls on the Cell phone much less attractive. I totaly agree that it makes it unattractive, but in no way does't it make the person calling me hesitate, so I can realy keep in touch. And so your spending level is dictated to you buy people that want to call you, at the whim of another (so to speak) Not really, but lets say that yes, the bottom line is that compare the same amount of minutes from your cell phone and landline with an american, and whoops you overpaid. All because of the call you make to cell phones. what about unlimited nights and weekends completely free that most providers give you here. What about the fact that even when you do pay for the incoming it costs around $.05 a minute? How about just not having to pay for incoming calls at all, that sounds much better. It makes being in touch easier and cheaper. Maybe, it makes it easier for the receiver but not for the one making the call. And it is the one that _chooses_ to make the call that make the decision to spend the money. Who's money should they be able
[Asterisk-Users] Panasonic KX-TD1232
If anyone has any experience with a Panasonic KX-TD1232 phone system, I would really like to talk to you for a few minutes. I have asterisk connected to a Panasonic system via FXS - CO ports. Im trying to get the Panasonic configured so that if someone dials a number (9) while Intercom is selected, it will select a line in the correct trunk group (Asterisk lines, rather than PSTN lines), then the user can finish dialing the asterisk extension. Any ideas? Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bill seconds [so far off topic that it has become a singularity]
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 June 2005 12:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bill seconds I showed you that your link to a mob called cucumber was not helpful to you or anyone else. Their pricing is fure fiction as far as .au telco pricing is concerned. Really pure fiction? Mob? let see: http://www.tel3advantage.com/rates.aspx?AgentNumber=036333CID=124 $.03 to regulare, and $.17 to mobile (more than 5 times as much) http://www.packet8.net/about/international.asp again $.03 to regular, and $.23 cents to mobile more than 7 times as much http://www.broadvoice.com/rateplans_international_li.html $.02 to regular, and $.18 to mobile 9 times as much http://www.voicepulse.com/plans/InternationalRates.aspx $.06 to regular and $.26 to mobile, that makes more than 4 times as much. anyhow to show you that cucumber is not the most expensive one: http://www22.verizon.com/ForYourHome/sas/sas_con_LongDescription.aspx $1.30 to australia here is one thats even better: http://www22.verizon.com/ForYourHome/sas/sas_basicinternationa lcallingcardrates.aspx Here is another Verizon rate: http://www22.verizon.com/ForYourhome/voip/CallingRates.aspx Don't ask me why the difference, but I promise you they don't even know. Fantastic, but not a single .au telco among them... Your telcos may not get great rates taling to our mobiles, but so what? I already included the link where it showed it costs more to call a cell phone. As for the the 10 times figure I made a mistake (since I was still under the impression that it costs only $.039 to call australia landline) and make it 4+ times as much (7 cents to landline and 30 to cell, that makes; 30/7=4+2/7 times as much as to a landline). That's what happens when you pull figures out of the air. chuckle Really out of the air? Yep. the interesting part here is that you know better than me that a huge chunk of your monthly phone bill (not your cell phone) goes towards phone calls made to mobile phones, Really? I have already told you that calls from my land line to my mobiles are free, what part of that didn't you understand? which is something that in the states doesn't exist, and still you argue that it doesn't cost you, and you divert this argument about what some company charges to Australia. Huh? What are you taking about? In an avarage month every American can tell you EXACTLY how much they are GOING to pay for their cellphone that month, and in most cases it is not a lot based on the minutes used. Ditto for .au However in places like Australia that you pay for your cell phone when calling from your home phone, there is no way of telling how much it is costing you since it costs you sometimes as much as 9 times as much to call a cell phone. *Sigh* I pay _exactly_ $0.00 each month to call my mobiles regardless of the number of calls, however you would have to pay to call _my_ mobiles, its called preselection, and it's a feature of my telco. for the 5 mobiles that I own, (my family members) the calls between them and my land lines are free. You already mentioned that (see below) that is NOT the argument. Because basic math teaches us that 2 negatives cancel each other, and I told you that the same is available in the states, so this argument is negated with the exact same argument that I have, and that is that I don't have to pay to ANY customer that is in the same network that I am (currently SprintPCS) nor does he pay for the incoming. So far all you have is only 5, and in the states I get about 30 Million phone numbers that I can call for free UNLIMITED (besides for nights and weekends that are completely free), so if you want this is another one for me. Kewl! Its tit-for-tat time :D Again, as the originator of the call I get to choose the amount I spend. Don't you see how they succeeded in making you believe that your cell phone is cheaper? I told you that none Amercians might not understand this. :) Yeah, I see how _some_ americans don't get it. when I don't originate the call, however in .us if you get called, you pay, that can easily cost you a heap of money that you can only control by switching the phone off, and where is the point in that? Really?? cost you a heap of money? only by swithcing the phone off? what ever happened to not picking up? Ok, there is that, so long as you take time to determine whether you recognise the number etc It does however make rec'ving calls on the Cell phone much less attractive. I totaly agree that it makes it unattractive, but in no way does't it make the person
[Asterisk-Users] Panasonic KX-TD1232
I can help you I think. do you have the manuals for the Panasonic? Quoting Dan Morin [EMAIL PROTECTED]: If anyone has any experience with a Panasonic KX-TD1232 phone system, I would really like to talk to you for a few minutes. I have asterisk connected to a Panasonic system via FXS - CO ports. I'm trying to get the Panasonic configured so that if someone dials a number (9) while Intercom is selected, it will select a line in the correct trunk group (Asterisk lines, rather than PSTN lines), then the user can finish dialing the asterisk extension. Any ideas? Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 500 Sound Problem
Hi all, I've been messing around with the g729 codec in some phones I use and had made all phones use the codec for all calls for testing purposes. The problem is when I attempt to dial out on my Polycom IP 500 (test happens to be calling my cell phone) I can only hear sound coming one way, I recieve nothing from one user, justsilence, yet I can talk one way perfectly. Now I tried the same call (forced g729) on my Cisco 7960 and had no problem connecting the call and having conversation heard both ways. I originally thought perhaps since I had only purchased one g729 channel license that it may cause such a problem, but then why does my 7960 connect with g729? I also had some problems with a SNOM 220 I had tried out as the audio was non-existant on both ends if I remember correctly, but I havent tried that one ina while so I may be incorrect. My primary concern is the Polycom anyway. I even had a buddy couble check through the gateway I am using that both calls went through as g729 and were connected (which obviously they were if any audio was heard). Is there something I am missing in the config on the Polycom 500? I am 99% sure the firmware is up-to-date... Thanks for any help, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] bluetooth audio and asterisk
This is what you want: http://www.voip-info.org/wiki-Asterisk+bounty+bluetooth+cell-phone+suppo rt Add money to the bounty -- maybe someone will pickup/continue what was started and abandoned. I think it's worth serious money. I know it is to me. Nice side-effect, for anyone interested -- in the US, all GSM phones are *required* to provide emergency dialing services. Wouldn't take much to keep a spare/used BT phone on hand to provide 911 calling even without a SIM card. -Original Message- From: trixter http://www.0xdecafbad.com [mailto:[EMAIL PROTECTED] Sent: Sunday, June 19, 2005 7:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bluetooth audio and asterisk Has anyone successfully used a standard bluetooth enabled system to connect to a standard bluetooth enabled mobile phone (not the bluetooth to FXS converters) to create an audio path for phone calls with asterisk, if so is there a writeup on what was done so that others can replicate this. What I am thinking is that via alsa/oss/whatever you should be able to use the bluetooth audio channel as a speaker and microphone to talk to a mobile. The catch is of course sending dialing information via bluetooth from within asterisk to cause this to properly occur. SMS could also be sent/received via bluetooth and converted to SIP IM for example thus giving more devices accessability to asterisk 'out of the box'. And given cost, many bluetooth dongles can be purchased for $30 USD or less, this would create a lower entry price for people to use this technology. Cellsocket.com for example is a phone-FXS adapter, they are $100. Someone this last week mentioned a bluetooth system that is similar, but afaik both devices do not allow for SMS to be trafficed via asterisk, and both cost more than a bluetooth dongle. I know that linux does support bluetooth via qualcomms BT API, and I would imagine that other systems also support this. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bill seconds [so far off topic that it hasbecome a singularity]
Who cares? MY cell phone is $0.00 per minute incoming AND outgoing... I can talk 24hrs a day to anyone in the United States or Canada, Wireless or landline, and my phone bill is going to be exactly the same every month. I'm sure there are good and bad cell phone plans in Australia AND in the UK. Why the pissing contest over one country being better/worse??? As far as not being able to control my cell phone bill because of incoming callers, isn't that what caller ID is for?? Not to mention that it IS still against the law to make telemarketing calls to cell phones in the US, though it's becoming harder to distinguish cell phone numbers from landline numbers, with number portability. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
