[Asterisk-Users] Auto-dial from webpage
I want to create a CGI that will allow me to make a call when a user clicks on a URL in a webpage. I believe I need to create a file in /var/spool/asterisk/outgoing that defines the number I want to call and the phone I want to connect it to but I see no way to define the phone number I want to dial in the /usr/src/asterisk/sample.call file I see mentioned in other posts. Is it possible to do what I want? Am I even looking in the right direction? elrod -- Mark Elrod Vindicia, Inc. 2755 Campus Drive, Suite 240 San Mateo, California 94403 Email: [EMAIL PROTECTED] Cell: 650-483-5763 Main: 650-292-2409 Fax: 650-345-1165 Web: http://www.vindicia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc John - I've discussed some of these with you, but here are a few for consumption and comments by the list: 8 - Encryption. * will probably support it as soon as some reasonable handsets support it. Grandstream should be the initiator in this process, with SRTP or some other RFC-approved method for delivering crypto SIP audio channels. Every business I talk to lists this on their priority chart for VoIP, and there are NO ANSWERS right now from major SIP handset vendors as far as voice crypto goes. I'm starting to think that it's a conspiracy. 8 - The TFTP configuration nonsense has been discussed. This needs to turn into an openly documented standard. Proprietary standards are useless, and all of them die eventually - why prolong the agony on your customers? 5 - Weight. Phone should weigh more. I'm constantly pulling it across the table with only the slightest stretching of the phone cord. 6 - Tilting display. Display should tilt up so I can actually read it. OR: base that tilts the whole phone up about 45 degrees (note: if this is the case, the weight issue really needs to be resolved) 9 - Buttons. The 102 model I have absolutely SUCKS as far as the buttons go. I have to pretty much press them like manual typewriter keys to get them to work. Any lateral force causes them to bind up. 10 - button response. Even when I _do_ manage to press the keys firmly enough, if I type too fast the keystrokes are lost. This is really, REALLY annoying. Button response needs to be sped up significantly. I almost always have to dial every number two or three times, or slow down to one button every second. Thus, I use my Cisco phones and leave the grandstream to gather dust. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto-dial from webpage
I want to create a CGI that will allow me to make a call when a user clicks on a URL in a webpage. I believe I need to create a file in /var/spool/asterisk/outgoing that defines the number I want to call and the phone I want to connect it to but I see no way to define the phone number I want to dial in the /usr/src/asterisk/sample.call file I see mentioned in other posts. Is it possible to do what I want? Am I even looking in the right direction? elrod -- Mark Elrod Vindicia, Inc. 2755 Campus Drive, Suite 240 San Mateo, California 94403 Email: [EMAIL PROTECTED] Cell: 650-483-5763 Main: 650-292-2409 Fax: 650-345-1165 Web: http://www.vindicia.com Mark - Hello! To create a call to 14109850123 on an analog channel in group 2 and then connect it to the hypothetical extension 84 (which would map to 84,1,Dial(SIP/84) ) inside your network, here's the file you'd create in /var/spool/asterisk/outgoing.call: # # Create the call on group 2 dial lines and set up # some re-try timers # Channel: Zap/g2/14109850123 MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that your local extensions are kept in the # context called [extensions] # Context: extensions Extension: 84 Priority: 1 I would strongly suggest that you create this file elsewhere, and copy it in after you're done creating it. Asterisk is very aggressive in grabbing these files, and if you're still creating it when it grabs the file, you'll get errors, so best to create first and then copy in to the outgoing directory all at once. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speed for meetme/asterisk?
Hello all, I'm planning to setup a linux box + asterisk as a h.323 conference bridge. My goal is to allow incoming h.323 calls with a voice menu that says welcome, and enter 1 thru 4 for public rooms or enter 5 then code for private rooms, or such. How much CPU power will Asterisk require to run say 16 participants in a bridge? Are we talking Pentium III 500 or AMD Athlon 2800? Anyone see any problems with my desired setup? I saw the notes about the zaptel phantom driver and USB for timing. Thanks in advance. -- Ethan / 757.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
John Brown (CV) wrote: Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. 1. More volume out of the speakerphone, and better range of the headset volume. I guess it would be sort of out there but if it were possible to separately adjust them that would be boss. To get the speakerphone to even be heard whilst hunching over it requires full volume. But then if someone calls and I don't put it back down it blasts my ears off. 2. Support for lower-b/w codecs. My list would include iLBC, Speex, and GSM. 3. Announced (supervised? consultative?) Transfers. 4. IAX support, which would lead to better NAT support. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto-dial from webpage
John Todd wrote:
I want to create a CGI that will allow me to make a call when a user
clicks on a URL in a webpage. I believe I need to create a file in
/var/spool/asterisk/outgoing that defines the number I want to call and
the phone I want to connect it to but I see no way to define the phone
number I want to dial in the /usr/src/asterisk/sample.call file I see
mentioned in other posts. Is it possible to do what I want? Am I even
looking in the right direction?
elrod
--
Mark Elrod
Vindicia, Inc.
2755 Campus Drive, Suite 240
San Mateo, California 94403
Email: [EMAIL PROTECTED]
Cell: 650-483-5763
Main: 650-292-2409
Fax: 650-345-1165
Web: http://www.vindicia.com
Mark -
Hello! To create a call to 14109850123 on an analog channel in
group 2 and then connect it to the hypothetical extension 84 (which
would map to 84,1,Dial(SIP/84) ) inside your network, here's the file
you'd create in /var/spool/asterisk/outgoing.call:
#
# Create the call on group 2 dial lines and set up
# some re-try timers
#
Channel: Zap/g2/14109850123
MaxRetries: 2
RetryTime: 60
WaitTime: 30
#
# Assuming that your local extensions are kept in the
# context called [extensions]
#
Context: extensions
Extension: 84
Priority: 1
I would strongly suggest that you create this file elsewhere, and copy
it in after you're done creating it. Asterisk is very aggressive in
grabbing these files, and if you're still creating it when it grabs
the file, you'll get errors, so best to create first and then copy in
to the outgoing directory all at once.
JT
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From your example you would dial the outbound line and once connected
dial the internal extension.. does this not mean that the external
person would hear the phone rining when it was answered??
I do mine the other way around..
Channel: SIP/84
WaitTime: 30
Context: {the context that has your outbound dial string}
Extension: 914109850123 ;9 added to number as if call was being placed
from a phone
Priority: 1
This way the internal extension is dialed first and when answerd the
external number is dialed..
Later..
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[Asterisk-Users] Call pickup - Change shortcode
Hello Is it possible, (without hacking the source), to change the code for call pickup because my SIP gateways uses * key as End-Of-Dial. If I have to hack the source can somebody tell me where to look? Mvh Mickey Binder Comflex A/S Tlf.: 43997102 Ë^®+$RÇ«²f¢)à+-Ë^®+$RÇ«²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®
Re: [Asterisk-Users] Survey: Grandstream improvements.........
John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes Hi John, My biggest issue is a hardware issue and is the single biggest reason why I have not been able to sell the GS phones into a company and that is the 10Mbps ethernet ports.. I guess if you are using the 101 then its not much of an issue but the whole cost saving is to cut down on wiring costs so the only model we even look at it the 102.. I don't have a single client that runs 10Mbps ethernet in their offices anymore and to tell them that the phone will downgrade their network speed to 10Mbps puts them off the phone straight away.. Staying with hardware, the screen needs to be angled a little to make it easier to read and needs to support more digits, and the buttons need to be easier to press.. Here are my suggestions for firmware updates.. 10 - Support for open low bandwidth codecs, specifically iLBC and GSM. 10 - Consultative Transfer. 7 - A nice feature of the Snom phones is the ability to type in the number with the handset still down and then the number is dialed when the handset is lifted or the OK button is pressed. This way you can take as long as you like to dial a number.. GS have the send/dial button so this feature should not be hard to add.. Adding to this.. it would be nice to be able to go through the Called and Callers call logs with the handset down and then when on the number you want to dial just lift the handset.. 5- Config refresh, apply config settings (even some of then) without needing to reboot the phone. Mark on the config page which settings will require a reboot to take effect.. 3 - Show the text part of the CallerID..(Think this may be a hardware issue or limitation) Fianally hardware support.. I had a power supply go on one of my GS phones, I purchased that phone from GS's agent in the US before there was anywhere to buy the phones in the UK, I contacted GS and then appologised and asked for my address, I assumed that was so I could be sent a replacement, I sent them my address..Now months later I have sent follow up emails which never get a response and I still don't have a replacement power supply.. so maybe you can speak to the president about that too.. That should about do it.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VM2 and MySQL
Hi, Does anyone have any instructions or pointers to get VM2 to work with MySQL? I can't seem to find any docs or how-to's on this.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Mon, 20 Oct 2003, John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. It goes without saying that consultative transfer has to be a 10 and I am sure I am not alone in saying so. Other things are niceties, but when selling to business this is an expected basic minimum. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Mon, 20 Oct 2003, John Todd wrote: 9 - Buttons. The 102 model I have absolutely SUCKS as far as the buttons go. I have to pretty much press them like manual typewriter keys to get them to work. Any lateral force causes them to bind up. 10 - button response. Even when I _do_ manage to press the keys firmly enough, if I type too fast the keystrokes are lost. This is really, REALLY annoying. Button response needs to be sped up significantly. I almost always have to dial every number two or three times, or slow down to one button every second. Thus, I use my Cisco phones and leave the grandstream to gather dust. I found that the buttons didn't work very well and I had lots of repeated or missed digits, making it almost impossible to login to the voicemail. However when I moved from using RTP to SIP INFO the problem vanished. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 21 Oct 2003, rnc Info Lists wrote: 9 - ability to switch back and forth between speakerphone and handset The Grandstream seems to have a strange method of working when it comes to speakerphone. I would expect the speakerphone button to just switch on and off the speaker, however it doesn't. If during a call you switch on the speaker then if you press th speakerphone button again to switch it off it hangs up the phone. However if you put the phone down instead and then pick it up again the speaker goes off and the call remains connected. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Mon, 20 Oct 2003, John Todd wrote: 9 - Buttons. The 102 model I have absolutely SUCKS as far as the buttons go. I have to pretty much press them like manual typewriter keys to get them to work. Any lateral force causes them to bind up. 10 - button response. Even when I _do_ manage to press the keys firmly enough, if I type too fast the keystrokes are lost. This is really, REALLY annoying. Button response needs to be sped up significantly. I almost always have to dial every number two or three times, or slow down to one button every second. Thus, I use my Cisco phones and leave the grandstream to gather dust. I found that the buttons didn't work very well and I had lots of repeated or missed digits, making it almost impossible to login to the voicemail. However when I moved from using RTP to SIP INFO the problem vanished. Michael My issue is not the encoding of the digits into the data stream, but the ability of the device to recognize the keystrokes. I use INFO, as well, after the usual failed experiments with inband and RFC2833 encoding. It just seems like there is some hardware issue that is not fast enough to catch my key presses. This is even before the call is handed off to the proxy (initial dial) so it's not a data transfer problem... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
I don't have a single client that runs 10Mbps ethernet in their offices anymore and to tell them that the phone will downgrade their network speed to 10Mbps puts them off the phone straight away.. Hey WipeOut, Maybe I am missing something here but why would it downgrade their network speed to 10mbps, its very rare to find a 100bT switches these days that don't also support 10bT. In a switched ethernet network there would be no performance loss for the other ports !? * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call forwarding
Hi all, I would like to forward an incoming call to my mobile with the incoming callerID after ten seconds, I have tried with this: exten = 910,1,Dial,SIP/[EMAIL PROTECTED]|10 exten = 910,2,Dial,CAPI/526$(CALLERIDNUM):bmymobilenr|20 exten = 910,3,Voicemail,u226 exten = 910,102,Voicemail,b226 The second row is not right and I can´t get it to work, any idea´s? ---JanM--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 21 Oct 2003, Low, Adam wrote: Maybe I am missing something here but why would it downgrade their network speed to 10mbps, its very rare to find a 100bT switches these days that don't also support 10bT. In a switched ethernet network there would be no performance loss for the other ports !? The cable goes into the phone and then out of the phone into the computer. That switch in the phone is 10Mbit so the computer ends up on 10Mbit too. Perhaps the best way to avoid this is to join all the phones together since they are all 10Mbit anyway, so you will then just need one extra ethernet socket in the room for all the telephones. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call forwarding
JanM wrote: Hi all, I would like to forward an incoming call to my mobile with the incoming callerID after ten seconds, I have tried with this: exten = 910,1,Dial,SIP/[EMAIL PROTECTED]|10 exten = 910,2,Dial,CAPI/526$(CALLERIDNUM):bmymobilenr|20 exten = 910,3,Voicemail,u226 exten = 910,102,Voicemail,b226 The second row is not right and I can´t get it to work, any idea´s? ---JanM--- I think your telco has to allow you to set the callerID of outbound calls, and I think they will usually only let you set it to one of your own numbers.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Bah, I replied directly instead of to the list. :-( 1 = Nice to have some day 10 = Got to have it right now 10 - Fix SIP disconnection problem 9 - Ringtones (downloadable?) 8 - ILBC 8 - MUCH MORE professional looking case (this includes dropping the four red LEDs beneath the white plastic face), maybe a nice black/gray/smoke matte plastic case, a wall mounting kit with a catch for the handset, etc. 7 - assisted transfer (I think that's what it's called) 6 - POE (12V-48V input range) 6 - 2.5mm headset jack 5 - integrated 100mbit switch ***capable of sustaining 100mbit*** 3 - IAX/IAX2 would be VERY nice 1 - downloadable codecs How's that for starters? Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
how do I prevent people from calling as soon as I restart the * server ? cos' this will result (I assume) in pri channels getting blocked. Because of those few calls that's taken during restart results in those few pri channels not to get properly restarted. I need something like 1~2 minutes calls free system when I restart *. --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 21 Oct 2003, Low, Adam wrote: Maybe I am missing something here but why would it downgrade their network speed to 10mbps, its very rare to find a 100bT switches these days that don't also support 10bT. In a switched ethernet network there would be no performance loss for the other ports !? The cable goes into the phone and then out of the phone into the computer. That switch in the phone is 10Mbit so the computer ends up on 10Mbit too. Perhaps the best way to avoid this is to join all the phones together since they are all 10Mbit anyway, so you will then just need one extra ethernet socket in the room for all the telephones. Michael Michael, How would you be able to connect all phones in a room to one socket? The Ethernet specificiation has a limit to the number of hubs/switches that can be inline. (or at least it used to). The only way I can see to connect all phones to one socket would be to daisy chain them. This would not be a good solution since: - all phones would use the same 10mbps segment, chances for collisions would be high - rules of Ethernet would be violated so even if it did work it may stop at any point with some other normally minor change. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Survey: Grandstream improvements.........
John Brown (CV) [EMAIL PROTECTED] said: Please keep in mind that adding new features take time to develop, test and such. 1. (8) Higher speakerphone volume - the current volume is inadequate; 2. (8) Lower DTMF volume - I usually use the volume at its highest setting (see 1.), the DTMF tones kill you. 3. (5) Documentation flyer you can present to end-users. 4. (1) Other low-bandwidth codecs, although I think that for a commercial application the $10 per channel for G.729 is not really that big an issue. Nice-to-have: IAX Hardware: - wallmount - 100Mbps ports -- Cees de Groot http://www.cdegroot.com [EMAIL PROTECTED] GnuPG 1024D/E0989E8B 0016 F679 F38D 5946 4ECD 1986 F303 937F E098 9E8B Cogito ergo evigilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 21 Oct 2003, rnc Info Lists wrote: Michael, How would you be able to connect all phones in a room to one socket? The Ethernet specificiation has a limit to the number of hubs/switches that can be inline. (or at least it used to). The only way I can see to connect all phones to one socket would be to daisy chain them. This would not be a good solution since: - all phones would use the same 10mbps segment, chances for collisions would be high - rules of Ethernet would be violated so even if it did work it may stop at any point with some other normally minor change. I defer to your knowledge in this area, but I would be interested to know what the limit is in terms of the number of devices that can be put inline. On the subject of collisions it seems to me that individual phone bandwidth use is relatively limited when compared to the 10Mbit/s available, so would the problem really be that substantial? Personally I currently have: Hub - Phone - Phone - Laptop No visible problems here, so certainly 3 phones in a line would seem to work. I suppose it all comes down to how many phones you put in a line. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
6 - 2.5mm headset jack 6.5 - when a headset is connected the ringer should NOT come through the headset... damn that is annoying on softphones... Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On 21/10/2003 11:14, Andrew Kohlsmith wrote: [...] 6 - POE (12V-48V input range) [...] 5 - integrated 100mbit switch ***capable of sustaining 100mbit*** [...] +1 on both of these points. The power brick is cheap and nasty. POE would be a huge plus. A 100mb bridge would make the phone a lot more attractive in an office full of cables. I'd also add my voice to the request for a better speakerphone. The dialtone comes out loud and clear but everything else is too muted. If I up the volume to hear calls, then the dialtone becomes deafening - as does the handset when used. I'm less concerned about the codecs as I'm happy to use ULAW/ALAW on the internal network and have Asterisk transcode to something else for external calls. There should be a way of locking the menu button, as it is too easy to muck with the settings. For central configuration, the cfg.txt file format would be nice, but is still a pain. Ideally I'd like to be able to configure the phone via DHCP extensions. That would be ideal as I can configure the lease time to manage how frequently the phones update and I can centralise the configuration with the IP details. Jonathan -- Jonathan Hogg Director, Technology Seventh Wave Systems Ltd. 4-14 Tabernacle Street London EC2A 4LU Telephone: +44 20 7074 0423 http://www.seventh-wave-systems.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
+1 on both of these points. The power brick is cheap and nasty. POE would be a huge plus. A 100mb bridge would make the phone a lot more attractive in an office full of cables. I specifically stated a wide POE range because let's face it, with the power requirements that phone has, a wide-input-range DC-DC converter is _peanuts_, especially if you've already got a tiny switchmode converter for line power. A very wide range on POE input makes it easy to mix and match phones too. Hell if you've got a switcher already, you can make it autosense polarity too. Don't pull a Cisco. Don't try and lock your users in to one brand of switches. As for the 100mbit switch -- again I was very specific here -- don't throw on one of those $0.25 100 mbit switch chips that can only sustain about 1MB/sec -- I put in a 100mbit switched network to achieve 11MB/sec sustained, not burst. A two-port switch capable of full sustained network speed shouldn't be expensive and can really be a big marketing feature. We won't screw your network speeds kind of thing. :-) I'd also add my voice to the request for a better speakerphone. The dialtone comes out loud and clear but everything else is too muted. If I up the volume to hear calls, then the dialtone becomes deafening - as does the handset when used. Speakerphone is a big deal with me too. For central configuration, the cfg.txt file format would be nice, but is still a pain. Ideally I'd like to be able to configure the phone via DHCP extensions. That would be ideal as I can configure the lease time to manage how frequently the phones update and I can centralise the configuration with the IP details. Why not specify a TFTP server/config filename via DHCP? It's already standard and would work very well. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
I love to have on my GS, GSM codec, scale = 10 - Original Message - From: rnc Info Lists [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 21, 2003 1:48 AM Subject: Re: [Asterisk-Users] Survey: Grandstream improvements. 7 - Ringer volume control 4 - plug in module of user programmable buttons for frequently called numbers. Not everyone would need this so being able to add as an optional module would keep the base phone cost effective. 9 - ability to switch back and forth between speakerphone and handset 7 - message waiting light under the message button. The LCD light blinking is nice but is not easy to see when the room is well lit. 4 - headset jack Thanks for taking the survey. You might also encourage David to have his folks actively participate in the lists. I mentioned it to him before and his reason for not having a more active presence was to avoid the appearance of being commercial on the lists. Personally, I think that it would help to build a better relationship between his technical folks and their userbase. Robert Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ 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] how to escape #
Yes, just press pound twice. Mark On Mon, 20 Oct 2003, Louis-David Mitterrand wrote: Hi, This morning I found myself stumped when a remote interactive system asked me to enter some identification followed by the # key, and my local Asterisk interrupted with Transfer?. Is there a way to escape the pound key, short of disabling transfers? Cheers, -- Make it idiot proof, and somebody will make a better idiot. ___ 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] Survey: Grandstream improvements.........
Andrew Kohlsmith wrote: Why not specify a TFTP server/config filename via DHCP? It's already standard and would work very well. This would need to be optional, what if a phone was deployed remotely where you have no control over the DHCP.. then you would need to specify the config file location or statically set the config.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
Hi! I defer to your knowledge in this area, but I would be interested to know what the limit is in terms of the number of devices that can be put inline. Correct me if I am wrong: 5 switches on 10 Mbit/s 2 switches on 100 Mbit/s (for the same segment) Note: Switches slow down your network... cable length matters as well, of course. Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieve_?_from_mysql.pl files??
I know I am posting to myself all the time here but as i didnt find any info on this when I was looking it may help others.. I have just been playing with the retrieve_sip_from_mysql.pl.. Some notes.. You must create an entry with the keyword account and the value will be what you want between the [] otherwise it will ignore any other parameters and not create the entry.. It does not create the [general] section of the sip.conf so I guess you will have to add that manually (anyone got any comments on this?).. By default it wants to create a file called sip_additional.conf.. Does Asterisk look for a file named sip_additional.conf when it loads? or do you have to merge the contents of sip.conf and sip_additional.conf?? Later.. WipeOut wrote: Hi, I was just taking a look at the source code and noticed two files.. retrieve_extensions_from_mysql.pl and retrieve_sip_conf_from_mysql.pl Its pretty obvious what these two files do, but info about them is a little scarce.. Is anyone using these scripts and could give me any details on them? Is there a similar script for voicemail.conf floating about? Thanks, 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] Survey: Grandstream improvements.........
Philipp von Klitzing wrote: Hi! Correct me if I am wrong: 5 switches on 10 Mbit/s 2 switches on 100 Mbit/s (for the same segment) Note: Switches slow down your network... cable length matters as well, of course. Philipp IIRC it was 5 HUB's on 10Mbps and 2 HUB's on 100Mbps, I seem to rememeber that when switches came along the rules got trashed and each manufacturer made their own rules.. But I could be wrong.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieve_?_from_mysql.pl files??
You'll want to #include it. This leaves the burden of the [general] and any static configs on sip.conf but allows the script to blindly write out from the database to sip_additional.conf in sip.conf: #include sip_additional.conf Steve On Tue, 21 Oct 2003, WipeOut wrote: I know I am posting to myself all the time here but as i didnt find any info on this when I was looking it may help others.. I have just been playing with the retrieve_sip_from_mysql.pl.. Some notes.. You must create an entry with the keyword account and the value will be what you want between the [] otherwise it will ignore any other parameters and not create the entry.. It does not create the [general] section of the sip.conf so I guess you will have to add that manually (anyone got any comments on this?).. By default it wants to create a file called sip_additional.conf.. Does Asterisk look for a file named sip_additional.conf when it loads? or do you have to merge the contents of sip.conf and sip_additional.conf?? Later.. WipeOut wrote: Hi, I was just taking a look at the source code and noticed two files.. retrieve_extensions_from_mysql.pl and retrieve_sip_conf_from_mysql.pl Its pretty obvious what these two files do, but info about them is a little scarce.. Is anyone using these scripts and could give me any details on them? Is there a similar script for voicemail.conf floating about? Thanks, 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 ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup
Hi, Some calls I make trough my PSTN asterisk gateway just hangup after some minutes. Even if I'm using sip or iax. I have callprogress=no busydetect=no in my zapata.conf. Anyone help? Or tell me what to look at /var/log/asterisk/debug. I didn't find anything wrong. [endpoint]---iax or sip[asterisk]EMPSTN. As endpoint I had tested another asterisk box (with a FXS), ciscoATA, cisco1750 and cisco827. The problem is the same with all. Eduardo debug when I call a PSTN number from the ATA186: == Oct 17 19:20:02 DEBUG[1605650]: File channel.c, Line 2210 (ast_channel_bridge): Didn't get a frame from channel: SIP/atasuporte-1413 Oct 17 19:20:02 DEBUG[1605650]: File channel.c, Line 2278 (ast_channel_bridge): Bridge stops bridging channels SIP/atasuporte-1413 and Zap/1-1 Oct 17 19:20:02 DEBUG[1605650]: File chan_zap.c, Line 1593 (zt_hangup): Hangup: channel: 1 index = 0, normal = 17, callwait = -1, thirdcall = -1 Oct 17 19:20:02 DEBUG[1605650]: File chan_zap.c, Line 1033 (zt_disable_ec): disabled echo cancellation on channel 1 Oct 17 19:20:02 DEBUG[1605650]: File chan_zap.c, Line 1960 (zt_setoption): Set option TDD MODE, value: OFF(0) on Zap/1-1 Oct 17 19:20:02 DEBUG[1605650]: File chan_zap.c, Line 992 (update_conf): Updated conferencing on 1, with 0 conference users Oct 17 19:20:02 DEBUG[1605650]: File chan_sip.c, Line 985 (sip_hangup): find_user(atasuporte) Oct 17 19:20:04 DEBUG[147466]: File chan_zap.c, Line 1033 (zt_disable_ec): disabled echo cancellation on channel 1 debug when I hangup the ATA186 === Oct 17 19:40:25 DEBUG[278546]: File dsp.c, Line 1212 (ast_dsp_process): Requesting Hangup because the busy tone was detected on channel Zap/1-1 Oct 17 19:40:25 DEBUG[278546]: File channel.c, Line 2218 (ast_channel_bridge): Got a FRAME_CONTROL frame on channel Zap/1-1 Oct 17 19:40:25 DEBUG[278546]: File channel.c, Line 2278 (ast_channel_bridge): Bridge stops bridging channels SIP/atasuporte-4c18 and Zap/1-1 Oct 17 19:40:25 DEBUG[278546]: File chan_zap.c, Line 1593 (zt_hangup): Hangup: channel: 1 index = 0, normal = 17, callwait = -1, thirdcall = -1 Oct 17 19:40:25 DEBUG[278546]: File chan_zap.c, Line 1033 (zt_disable_ec): disabled echo cancellation on channel 1 Oct 17 19:40:25 DEBUG[278546]: File chan_zap.c, Line 1960 (zt_setoption): Set option TDD MODE, value: OFF(0) on Zap/1-1 Oct 17 19:40:25 DEBUG[278546]: File chan_zap.c, Line 992 (update_conf): Updated conferencing on 1, with 0 conference users Oct 17 19:40:25 DEBUG[278546]: File chan_sip.c, Line 985 (sip_hangup): find_user(atasuporte) Oct 17 19:40:25 DEBUG[81926]: File chan_sip.c, Line 544 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Oct 17 19:40:28 DEBUG[147466]: File chan_zap.c, Line 1033 (zt_disable_ec): disabled echo cancellation on channel 1 = ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieve_?_from_mysql.pl files??
Are you manually updating the mySQL tables or do you have a web app. to do that? Robert Steve Creel wrote: You'll want to #include it. This leaves the burden of the [general] and any static configs on sip.conf but allows the script to blindly write out from the database to sip_additional.conf in sip.conf: #include sip_additional.conf Steve Excellent, Thanks for that.. I didn't know there was an include command.. Do you know if include is available in other .conf files eg extensions.conf?? 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] retrieve_?_from_mysql.pl files??
It's documented somewhere for extensions.conf, and I was delighted to see that it is a function of the config parser, so yes - it's available in the other files. On Tue, 21 Oct 2003, WipeOut wrote: Steve Creel wrote: You'll want to #include it. This leaves the burden of the [general] and any static configs on sip.conf but allows the script to blindly write out from the database to sip_additional.conf in sip.conf: #include sip_additional.conf Steve Excellent, Thanks for that.. I didn't know there was an include command.. Do you know if include is available in other .conf files eg extensions.conf?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieve_?_from_mysql.pl files??
rnc Info Lists wrote: Are you manually updating the mySQL tables or do you have a web app. to do that? Robert I am manually updating the DB as I have just started playing with the files today.. but using phpmyadmin its really easy to do.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
It goes without saying that consultative transfer has to be a 10 and I am sure I am not alone in saying so. Other things are niceties, but when selling to business this is an expected basic minimum. I fully agree with that. on my list, 'supervised transfer' is the more (software) feature needed. then goes ringtones and at least gsm codec On the hardware point of view : real 100Mbits interface, heavier case ? matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
John, I second Brian's comments. After setting up 20 GS phones using their somewhat odd web interface, I would really appreciate a more rational provisioning system for small to medium installations. I would add the following: cfgEveryone.txt:Generic setup for all phones. Read first - and overridden by cfgMACADDRESS.txt:the specific setup items for each phone. Note, that there already seems to be a config file format (undocumented). If this is true, GS should at least publish the format and let the OS community have a go at a configurator. Also, the daylight savings option will ultimately need to be fixed to include date recognition. The current setup requires that you log into every phone twice each year to turn the option on/off. For installations with a large number of phones, this is going to be a real headache. And, the speakerphone button needs to be fixed. It works now almost perfectly. The only glitch is when you are on the speakerphone and want to switch back to the handset. If the handset is on the cradle, picking it up will transfer the call to the headset from the speakerphone. However, if you have the handset off hook already and press the speakerphone button expecting to transfer back to the handset, you are disconnected. The documentation states that it is a toggle. It isn't. The workaround is to press the on-hook button momentarily and you are switched back to the handset. Nevertheless, the speakerphone button should not hang up the line unless the receiver is already on hook. Finally, the documentation for IP QOS, VLAN Tag and Dialplan need to be expanded/included. Stephen R. Besch John, I want the tftp configs done like cfgMACADDRESS.txt or compile them into a binary form like the ATA's use. And stop trying to rip us for the GAPS system. WHAT A RIP. It makes cisco so worth the extra cash! Config refresh similar to the ATA.. refresh config every x seconds. bkw On Mon, 20 Oct 2003, John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ 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
[Asterisk-Users] unclear about IAX
Hello everyone: I have a few questions about IAX. As I understand IAX is used for Inter-* communications. 1. How do the asterisk boxes communicate? Is it over IP or some other mechanism (T1 trunk?) 2. If I have a scenario where I'd like to use * as a PBX in a Small to medium enterprise and at the same time use another Asterisk box at a service providers office where it connects to a PSTN and the internet. Would such a setup work with IAX? 3. Is there some documentation about setting up IAX and different deployment scenarios? Thanks for your help. Deepak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI problem (crash) in RH9
Hi Thank you everyone for your help. I got Asterisk to stop crashing by installing a 2.4.22 kernel. Best regards, Ívar Ragnarsson -Original Message- From: Michael T Farnworth [mailto:[EMAIL PROTECTED] Sent: 17. október 2003 22:21 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] AGI problem (crash) in RH9 On Fri, 17 Oct 2003, mattf wrote: Redhat has EVERYTHING set to LANG=UTF-8 and it screws up all sorts of perl stuff, and several other pre-written programs in other languages too It is a pain, and it even breaks man pages and all sorts of other things in my experience. I recommend disabling the UTF-8 default by editing: /etc/sysconfig/i18n The top line probably reads something like: LANG=en_US.UTF-8 change it to: LANG=en_US or en_GB if you are a UK person. You probably need to reboot after doing that. Michael MATT--- ___ 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] CallerID Screening Prohibit
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, How can I check if (i.e.) my provider is requesting me to hide the callerid? I.e. (Telco)E1/PRI---Zap(Asterisk1)IAX---IAX(Asterisk2)SIP---EP Now, if a call comes from the Telco with CLI screening prohobited to Asterisk1, where the call is forwarded using Dial() via IAX to Asterisk2 and then (also using Dial()) on to a SIP endpoint, how do I make sure the CLI is kept hidden all the way? Currently I'm using my own Dial() patch which set the restrictcid flag on demand. Could it be extracted from an incoming call somehow? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQE/lURd2TEAILET3McRAkBmAJ4vva/kqMFLFiRHRUHyIegh4Ml7AACfVcK9 zbrwPYjgTK0EAoJ/iiT1cIs= =+Lz5 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto-dial from webpage
Channel: Zap/g2/14109850123
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: extensions
Extension: 84
Priority: 1
JT
From your example you would dial the outbound line and once connected
dial the internal extension.. does this not mean that the external
person would hear the phone rining when it was answered??
I do mine the other way around..
Channel: SIP/84
WaitTime: 30
Context: {the context that has your outbound dial string}
Extension: 914109850123 ;9 added to number as if call was being placed
from a phone
Priority: 1
This way the internal extension is dialed first and when answerd the
external number is dialed..
Elrod,
Though it may look like the extension is dialed first, it isn't. The
order of that file doesn't mater and * will always dial the extension
first. Give John's example a try and it should work.
Here's some tips:
- the file name doesn't mater
- set up a file in another directory then just copy the file to
/var/spool/asterisk/outgoing and * will process and delete it. This way
you can easily modify your test file
- Careful which context you use in /etc/asterisk/extensions.conf to make
sure it's correct.
-adj
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RE: [Asterisk-Users] Survey: Grandstream improvements.........
I have a Nortel phone on my desk right now. IF the handset is picked up and you press the speaker button, it does not hang up but switches back to the handset instead. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Robert Hajime Lanning Sent: Tuesday, October 21, 2003 10:26 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Survey: Grandstream improvements. quote who=Michael T Farnworth On Tue, 21 Oct 2003, rnc Info Lists wrote: 9 - ability to switch back and forth between speakerphone and handset The Grandstream seems to have a strange method of working when it comes to speakerphone. I would expect the speakerphone button to just switch on and off the speaker, however it doesn't. If during a call you switch on the speaker then if you press th speakerphone button again to switch it off it hangs up the phone. However if you put the phone down instead and then pick it up again the speaker goes off and the call remains connected. I never had this problem. As all the PBX phones (currently NorTel Meridian) that I have used work that way. (Speaker button turns on the speaker, use hook button to switch back to handset.) -- END OF LINE ___ 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
(Ethernet issues) RE: [Asterisk-Users] Survey: Grandstream improvements.........
quote who=Michael T Farnworth On Tue, 21 Oct 2003, rnc Info Lists wrote: Michael, How would you be able to connect all phones in a room to one socket? The Ethernet specificiation has a limit to the number of hubs/switches that can be inline. (or at least it used to). The only way I can see to connect all phones to one socket would be to daisy chain them. This would not be a good solution since: - all phones would use the same 10mbps segment, chances for collisions would be high - rules of Ethernet would be violated so even if it did work it may stop at any point with some other normally minor change. I defer to your knowledge in this area, but I would be interested to know what the limit is in terms of the number of devices that can be put inline. On the subject of collisions it seems to me that individual phone bandwidth use is relatively limited when compared to the 10Mbit/s available, so would the problem really be that substantial? Personally I currently have: Hub - Phone - Phone - Laptop No visible problems here, so certainly 3 phones in a line would seem to work. I suppose it all comes down to how many phones you put in a line. Michael Too many switches/hubs will cause late collisions. Late collisions are ethernet collisions that happen after the transmitting station has finished transmitting. If it is a store and forward switch, then the switch can retransmit on collision, otherwise the packet is completely lost. This is the same reason why an ethernet cable cannot be over 300 feet. The first bit of the ethernet frame must get to the farthest node in an ethernet segment before the last bit is transmitted by the originating station. This length is based on speed one bit takes to span the distance and the minimum ethernet frame size (64 bytes). Currently the limit is 5 non-store and forward switches/hubs. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandsteam to support iLBC
Since quite a few people in the Grandstream improvements. thread have requested support for other low bandwidth codecs. I thought I would post this link. http://www.globalipsound.com/newsroom/releases.php?newsID=46 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A software FAX modem
Hi, I did say this was a first test release :-) I can't be held responsible for libtiff being empty on your machine, but they other issues are my fault. I have put a new tarball up, which should need nothing more than libtiff for the library to work. All the other libraries you had issues with are only needed by the test programs. ./configure will now only set things up to build the library. ./configure --enable-doc --enable-tests --enable-itutests will set things up to be everything. This newer tarball also has improvements in the software. The V.29 modem now works at all three speeds - 9600, 7200, and 4800bps. Previously I had only debugged the 9600bps mode. FAX doesn't actually use the 4800bps mode, but I thought I ought to fully implement the spec. (thats another way of saying it was easy to do after getting 9600 and 7200 working :-) ). I also fixed something that could cause crashes if you tried to send a non-existant TIFF file. I'm sorry if I disappointed the early adopters, but it *will* get better. Regards, Steve Florian Overkamp wrote: Hi, Citeren Steve Underwood [EMAIL PROTECTED]: If it doesn't work for you, don't be too surprised. Feed back anything you find, and lets try to make things better. I suspect, from experience and things I have read on the web, that a lot of fax machines do not follow the standards very well. In that case, a number of tweaks are probably needed before this new software is adequately tolerant of the behaviour of real world machines. First off, let me start by saying I think this is a great new step that is greatly appreciated (at least by me) toward a complete telephony platform. Second off, I just tried to build and install. Some comments up till now: - The compilation process asks for libaudiofile headerfiles (-dev package) - this was not default on my box. Should be added to documentation I guess :) - On my Debian box libtiff is an empty package (sucks) so I downloaded the tiff package source code. Installation here sucks once more: the mentioned tiffiop.h is not installed in /usr/local/include as I suppose it should, same goes for several other header files. Easiest was to just point the Makefile in your src/ tree toward the libtiff source. - The linker tries to access fftw (Fourier libraries). Not default installed on my system, should probably be added to documentation - The linker tries to access unicall. What is this ? Not installed on my system and no candidates on my searchlist (apt-get and a quick google search). How to continue ? Thanks, and I hope to continue this adventure soon :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
I never had this problem. As all the PBX phones (currently NorTel Meridian) that I have used work that way. (Speaker button turns on the speaker, use hook button to switch back to handset.) Agreed. One thing that would be nice though is to emulate the meridian's use of the handsfree button as a mic mute toggle when in handsfree mode. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
I have a Nortel phone on my desk right now. IF the handset is picked up and you press the speaker button, it does not hang up but switches back to the handset instead. Not with my Meridian system. Just tested to verify: handset onhook + handsfree/mute pressed: handsfree (goes off-hook) handset offhook + handsfree/mute pressed: handsfree handsfree offhook + handsfree/mute pressed: mic mute toggle In all cases, to get back to handset use you must toggle hook switch. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Even Newer Patch to app_queue with skillbased strategy
and the bits 1,2,4 For the queue skillmask just keep multing the number by 2 1 = sales 2 = tech level 1 4 = tech level 2 8 = tech level 3 16 = advanced problems 32 = coperate to allow a queue member to be allowed to take the call just add up all the numbers that go with his skills and set that as his skillmaskI suggest looking at ast_get_group() for an easy way to do what you'retrying to do here with the group numbers. Your ideas are great though :)Mark I did encounter the groups but I was encountering some difficulty with the metric when I added a group into a queue it always seemed to win the metric battle despite the penalty. Something I have been playing with is taking the penalty and using it to rank the queue members by regenerating queues.conf every 1 hour with every agent defined seperatly and each with a different penalty so , in the case of sales, using the cdr and some carefully plannedrelations between sales records and a database I would calculate the efficiency of the sales team every 1 hour and redraw the queue moving the best performers to the top. My motivation for the skillbased routing comes more from a tech support requirement than a sales one. My intention is to make a database of agents with the various skills they possess all wrapped up into that 1 numeric representation. My examples would have better suited me if they demonstrated this better 1 = email expert 2 = unix expert 4= windows expert (huh?) 7= hardware specialist Then when a call comes in and after a few questions you determine the call requires the attention of someone who understandsunix issues Onewould pass thevalue2 as the skillmask allowing any agent with the 'unix' bit set (a.k.a. 2) to take the call. Also I want to implement dynamic queue addition using this concept so an AGI can be used to query your agent id then make several calls to AddQueueMember and report your current skillset via the skillmask parameter. I know asterisk has an app for talking to pg and so forth but what I like the most about it is it's similarity to 'legos' so I try and follow that mentality where you dont make features for it you make interfaces to hang your features. just like lego only provides you with a pile of bricks and a few specialized ones such as motor lights etc.. Do you Yahoo!? The New Yahoo! Shopping - with improved product search
[Asterisk-Users] compile problems with suse 8.2
Can anybody give me some directions how i could compile Asterisk for SuSE 8.2 of 9.0. I have a lot of problems compiling. I have installed the readline-dev ncurses package etc. Or maybe somebody has a rpm for SuSE available? With kind regards, Martin Temmink. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto-dial from webpage
Ashley Jones wrote:
Channel: Zap/g2/14109850123
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: extensions
Extension: 84
Priority: 1
JT
From your example you would dial the outbound line and once
connected dial the internal extension.. does this not mean that the
external person would hear the phone rining when it was answered??
I do mine the other way around..
Channel: SIP/84
WaitTime: 30
Context: {the context that has your outbound dial string}
Extension: 914109850123 ;9 added to number as if call was being
placed from a phone
Priority: 1
This way the internal extension is dialed first and when answerd the
external number is dialed..
Elrod,
Though it may look like the extension is dialed first, it isn't. The
order of that file doesn't mater and * will always dial the extension
first. Give John's example a try and it should work.
Here's some tips:
- the file name doesn't mater
- set up a file in another directory then just copy the file to
/var/spool/asterisk/outgoing and * will process and delete it. This
way you can easily modify your test file
- Careful which context you use in /etc/asterisk/extensions.conf to
make sure it's correct.
-adj
Ashley,
It would appear that the order IS important.. I have just tried a test
file with the Channel: entry being my Cell number and the
Context/Exten/Priority being my extension and it did as I expected, It
dialed my Cell phone fisrt and when I answered my Cell phone I heard
ringing as my extension on my dask started to ring..
Looks like it must be done in the order of source(extension) then
destination(external phone number)..
Later..
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Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Tuesday 21 October 2003 01:07, John Todd wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. 5 - Weight. Phone should weigh more. I'm constantly pulling it across the table with only the slightest stretching of the phone cord. I'd have to respectfully disagree. If this is really a problem I'd suggest taking advantage of the mounting bracket on the bottom and either attach the phone to the desk or attach a sheet of lead. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: (Ethernet issues) RE: [Asterisk-Users] Survey: Grandstream improvements.........
Personally, I just wire every jack the same way back to the patch panels, 4pr cat5 or better, terminated in an rj45. Back at the panel wire the blue pair to your analog telephony stuff, and the org/grn to your networking. Then if you plug in an rj11 you get a phone line, if you plug in a network cable that works too. Some would say this is wasteful of wire, but in reality the wire is the least part of the cost of a cabling installation. Labour far outweighs it. If you want a physical 10mb/sec subnet for your phones, easy, just patch the relevant jacks into that hub/switch, separated from the jacks used from your data network. There are also some ways to stretch this distance limit if you are careful, and limit the branching topology of the lan segment. This is the same reason why an ethernet cable cannot be over 300 feet. The first bit of the ethernet frame must get to the farthest node in an ethernet segment before the last bit is transmitted by the originating station. This length is based on speed one bit takes to span the distance and the minimum ethernet frame size (64 bytes). Currently the limit is 5 non-store and forward switches/hubs. -- END OF LINE ___ 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] No detection of Line Busy
On Tue, 2003-10-21 at 01:30, Herc wrote: Quoting Steven Critchfield [EMAIL PROTECTED]: Your problem basically comes from the fact that in the analog world, a busy signal is a audio only signaling. Asterisk could do this detection if it was directly touching the phones, or if it was connected digitally to the switch that touches the phones. I think there are some busydetect features you can turn on in the zapata.conf file, but they are one of the things that is prone to diconnecting a call early as you are trying to fuzzy match a pattern of sound at the beginning of the call. -- Steven Critchfield [EMAIL PROTECTED] Yes, we tried enabling busydetect in zapata.conf file. then, once the extension is busy, as u suggested, asterisk disconnects the line. in a situation like this, instead of disconnecting, cant we dial another number? don't u think we should have this fascility..??? I'm not arguing against this functionality. I'm just trying to explain what is currently possible with the functionality that is already implemented. Maybe the result of busydetect needs to be checked out. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto-dial from webpage
- Original Message -
From: WipeOut [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 21, 2003 3:32 AM
Subject: Re: [Asterisk-Users] Auto-dial from webpage
John Todd wrote:
I want to create a CGI that will allow me to make a call when a user
clicks on a URL in a webpage. I believe I need to create a file in
/var/spool/asterisk/outgoing that defines the number I want to call and
the phone I want to connect it to but I see no way to define the phone
number I want to dial in the /usr/src/asterisk/sample.call file I see
mentioned in other posts. Is it possible to do what I want? Am I even
looking in the right direction?
elrod
--
Mark Elrod
Vindicia, Inc.
2755 Campus Drive, Suite 240
San Mateo, California 94403
Email: [EMAIL PROTECTED]
Cell: 650-483-5763
Main: 650-292-2409
Fax: 650-345-1165
Web: http://www.vindicia.com
Mark -
Hello! To create a call to 14109850123 on an analog channel in
group 2 and then connect it to the hypothetical extension 84 (which
would map to 84,1,Dial(SIP/84) ) inside your network, here's the file
you'd create in /var/spool/asterisk/outgoing.call:
#
# Create the call on group 2 dial lines and set up
# some re-try timers
#
Channel: Zap/g2/14109850123
MaxRetries: 2
RetryTime: 60
WaitTime: 30
#
# Assuming that your local extensions are kept in the
# context called [extensions]
#
Context: extensions
Extension: 84
Priority: 1
I would strongly suggest that you create this file elsewhere, and copy
it in after you're done creating it. Asterisk is very aggressive in
grabbing these files, and if you're still creating it when it grabs
the file, you'll get errors, so best to create first and then copy in
to the outgoing directory all at once.
Creating this file elsewhere would help in keeping track of whether or not your script
created the file, and if it did so correctly.
Also, you would probably see results of these posts in your web log files, but having
a datestamped dir of log entries couldn't hurt.
JT
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From your example you would dial the outbound line and once connected
dial the internal extension.. does this not mean that the external
person would hear the phone rining when it was answered??
I do mine the other way around..
Channel: SIP/84
WaitTime: 30
Context: {the context that has your outbound dial string}
Extension: 914109850123 ;9 added to number as if call was being placed
from a phone
Priority: 1
This way the internal extension is dialed first and when answerd the
external number is dialed..
Later..
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-
Andrew Thompson
NetResults, Inc.
(910) 215-9991 x301
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[Asterisk-Users] Defragmenting mailboxes
Does anyone have a quick and dirty script for defragmenting mailboxes? i.e.: -rwx--1 root root80553 Oct 20 16:27 msg.gsm -rw-r--r--1 root root 218 Oct 20 16:27 msg.txt -rwx--1 root root 781164 Oct 20 16:27 msg.wav -rwx--1 root root79360 Oct 20 16:27 msg.WAV -rwx--1 root root 7260 Oct 20 17:48 msg0001.gsm -rw-r--r--1 root root 225 Oct 20 17:48 msg0001.txt -rwx--1 root root70444 Oct 20 17:48 msg0001.wav -rwx--1 root root 7211 Oct 20 17:48 msg0001.WAV -rwx--1 root root11220 Oct 21 08:13 msg0002.gsm -rw-r--r--1 root root 224 Oct 21 08:13 msg0002.txt -rwx--1 root root 108844 Oct 21 08:13 msg0002.wav -rwx--1 root root1 Oct 21 08:13 msg0002.WAV -rwx--1 root root37092 Oct 21 10:53 msg0003.gsm -rw-r--r--1 root root 226 Oct 21 10:53 msg0003.txt -rwx--1 root root 359724 Oct 21 10:53 msg0003.wav -rwx--1 root root36590 Oct 21 10:53 msg0003.WAV -rwx--1 root root 175791 Oct 10 14:56 msg0009.gsm -rw-r--r--1 root root 227 Oct 10 14:56 msg0009.txt -rwx--1 root root 1704684 Oct 10 14:56 msg0009.wav -rwx--1 root root 173156 Oct 10 14:56 msg0009.WAV -rwx--1 root root65340 Oct 10 17:03 msg0010.gsm -rw-r--r--1 root root 217 Oct 10 17:03 msg0010.txt -rwx--1 root root 633644 Oct 10 17:03 msg0010.wav -rwx--1 root root64410 Oct 10 17:03 msg0010.WAV Note the gap between 0003 and 0009. This is caused by a somewhat common situation, and it tends to bite us somewhat often. :-) If not, if I get a chance, I'll whip something up. -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] zhone z-plex 10
Is there a trick to getting into a zhone z-plex 10 through the serial interface? I tried using a couple of terminal programs the other day and didn't get a login. I have a port that quit working and it doesn't appear to be wiring related. I had another case like this the other day, and after fiddling with the wiring for a long time, I was ready to give up for the night. I restarted everything one last time for grins and the port came back up. Now I have a different port down and I am wondering if the z-plex is downing the port for some reason or other. Thanks, Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
We have a 10 and we need it yesterday (as well as many other people who don't even know it). We have a Bug report at GS. The problem is with STUN and changing IP Addresses. It happens like this: 1. Phone does a STUN query and registers fine. 2. If the public IP Address changes sometime later (like on a DSL line that disconnects and connects back), the phone will keep registering with the original IP address, and thus will fail to work properly. It apparently does not attempt further STUN queries for registration purposes. STUN isn't even needed nat=yes is all you need and it just works. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Beta testers for visual configuration tool for asterisk
Hi, thanks for you reply. I'll send you till the end of the week more info on how to download and use it. Best regards, Paulo Mannheimer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lal, Deepak (Contractor) Sent: sexta-feira, 17 de outubro de 2003 14:52 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Beta testers for visual configuration tool for asterisk Count me in too. -Original Message- From: sip [mailto:[EMAIL PROTECTED] Sent: Friday, October 17, 2003 1:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk count me in - Original Message - From: Paulo Mannheimer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 17, 2003 12:23 PM Subject: [Asterisk-Users] Beta testers for visual configuration tool for asterisk Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland doesn't crash because of a missing sound file. Beta users will have to download and install either a 16 Mb or a 4Mb Windows program, depending if you already have or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com ___ 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] A software FAX modem
I did say this was a first test release :-) I can't be held responsible for libtiff being empty on your machine, but they other issues are my fault. I have put a new tarball up, which should need nothing more than libtiff for the library to work. All the other libraries you had issues with are only needed by the test programs. ./configure will now only set things up to build the library. ./configure --enable-doc --enable-tests --enable-itutests will set things up to be everything. This newer tarball also has improvements in the software. The V.29 modem now works at all three speeds - 9600, 7200, and 4800bps. Previously I had only debugged the 9600bps mode. FAX doesn't actually use the 4800bps mode, but I thought I ought to fully implement the spec. (thats another way of saying it was easy to do after getting 9600 and 7200 working :-) ). I also fixed something that could cause crashes if you tried to send a non-existant TIFF file. I'm sorry if I disappointed the early adopters, but it *will* get better. We tried to use it witch our AVM Fritz!Card with chan_capi but asterisk always crashes after our fax-machines shows the ID of the soft-fax (12345678). Here's a backtrace: #0 0x417546a4 in TIFFWriteBufferSetup () from /usr/lib/libtiff.so.3 #1 0x417547fd in TIFFFlushData1 () from /usr/lib/libtiff.so.3 #2 0x4174019a in _TIFFFax3fillruns () from /usr/lib/libtiff.so.3 #3 0x417413d1 in _TIFFFax3fillruns () from /usr/lib/libtiff.so.3 #4 0x415f0270 in t4_rx_end_page (s=0x415fa0a4) at t4.c:330 #5 0x415e94e8 in key () from /usr/lib/asterisk/modules/app_rxfax.so #6 0x415ee062 in v29_rx () from #/usr/lib/asterisk/modules/app_rxfax.so #7 0x415ec485 in fax_rx_process () from #/usr/lib/asterisk/modules/app_rxfax.so #8 0x415e9068 in rxfax_exec (chan=0x8136550, data=0xbc7ff7b4) at #app_rxfax.c:183 #9 0x08060af0 in pbx_exec (c=0x8136550, app=0x8119248, #data=0xbc7ff7b4, newstack=1) at pbx.c:396 #10 0x08062ad3 in pbx_extension_helper (c=0x8136550, context=0x81366a8 #default, exten=0x813679c 3841, priority=1, callerid=0x80df9f8 #3843, action=1) at pbx.c:1151 #11 0x0806380d in ast_pbx_run (c=0x8136550) at pbx.c:1635 #12 0x0806988e in pbx_thread (data=0x8136550) at pbx.c:1856 #13 0x400310ba in pthread_start_thread () from /lib/libpthread.so.0 Hope that helps -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature
[Asterisk-Users] Iitter Buffer Settings
I'm trying to come up with good jitterbuffer related settings for my Asterisk boxes. I ran 4 pings for about 2 days from my main Asterisk server to remote Asterisk servers. During that time there were some large file uploads which caused the max rtt to be quite large. Here are the results: pktslossmin avg max mdev 132013 %0 70.36 78.13 1967.37 36.04 132013 %0 98.95 120.46 2419.24 111.26 132040 %0 33.82 42.88 1904.79 36.62 131919 %0 0.0047.04 1924.64 36.36 Using these numbers, and knowing that the max rtt will not happen very often how do the jitter settings below look? Does anyone have any recommendations to improve my call quality using the jitterbuffer? jitterbuffer=300 dropcount=1 maxjitterbuffer=500 maxexccessbuffer=20 -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Beta testers for visual configuration tool f or asterisk
Hi, thanks for you reply. I'll send you till the end of the week more info on how to download and use it. Best regards, Paulo Mannheimer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Wienecke Sent: sexta-feira, 17 de outubro de 2003 17:43 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool f or asterisk Am Freitag, 17. Oktober 2003 19:51 schrieb Lal, Deepak (Contractor): i am willing to assist also. mostly on weekends, i´ m afraid, but willing. Thomas W. Count me in too. -Original Message- From: sip [mailto:[EMAIL PROTECTED] Sent: Friday, October 17, 2003 1:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk count me in - Original Message - From: Paulo Mannheimer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 17, 2003 12:23 PM Subject: [Asterisk-Users] Beta testers for visual configuration tool for asterisk Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland doesn't crash because of a missing sound file. Beta users will have to download and install either a 16 Mb or a 4Mb Windows program, depending if you already have or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Beta testers for visual configuration tool for asterisk
Hi, thanks for you reply. I'll send you till the end of the week more info on how to download and use it. Best regards, Paulo Mannheimer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Coberly Sent: sábado, 18 de outubro de 2003 14:49 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk Hi, We would be interested in this project also. Paulo Mannheimer wrote: Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland doesn't crash because of a missing sound file. Beta users will have to download and install either a 16 Mb or a 4Mb Windows program, depending if you already have or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM ___ 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] Beta testers for visual configuration tool for asterisk
Hi, thanks for you reply. I'll send you till the end of the week more info on how to download and use it. Best regards, Paulo Mannheimer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Roberson Sent: sábado, 18 de outubro de 2003 01:21 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Beta testers for visual configuration tool for asterisk I would like to beta test this tool. :) Looks like it could be a good thing. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Mannheimer Sent: Friday, October 17, 2003 11:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Beta testers for visual configuration tool for asterisk Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland doesn't crash because of a missing sound file. Beta users will have to download and install either a 16 Mb or a 4Mb Windows program, depending if you already have or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 ___ 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] Survey: Grandstream improvements.........
My issue is not the encoding of the digits into the data stream, but the ability of the device to recognize the keystrokes. I use INFO, as well, after the usual failed experiments with inband and RFC2833 encoding. It just seems like there is some hardware issue that is not fast enough to catch my key presses. This is even before the call is handed off to the proxy (initial dial) so it's not a data transfer problem... I use RFC2833 and it works fine... as for switching to and from handset and speakerphone it can be done. press speaker phone ... your on speaker phone... hangup the handset.. then when you pick the handset back up you are on speaker phone. just an FYI bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
I alwasy laff at those DISCLAIMERS on email... funny they are at the bottom. bkw On Tue, 21 Oct 2003, Low, Adam wrote: I don't have a single client that runs 10Mbps ethernet in their offices anymore and to tell them that the phone will downgrade their network speed to 10Mbps puts them off the phone straight away.. Hey WipeOut, Maybe I am missing something here but why would it downgrade their network speed to 10mbps, its very rare to find a 100bT switches these days that don't also support 10bT. In a switched ethernet network there would be no performance loss for the other ports !? * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ 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] Defragmenting mailboxes
On Tue, 2003-10-21 at 09:44, Ryan Tucker wrote: Does anyone have a quick and dirty script for defragmenting mailboxes? [snip] Note the gap between 0003 and 0009. This is caused by a somewhat common situation, and it tends to bite us somewhat often. :-) If not, if I get a chance, I'll whip something up. -rt Yes, the gaps in the numbering get really annoying. Unfortunately, it's a little bit risky to go moving files around without stopping asterisk first, or you might just contribute to the problem instead of helping it. (For example, if someone is leaving a message while you are renumbering the files.) I think the best long-term solution is to [ask|beg|pay|coerce|convince] someone to fix the way voicemail messages are numbered to avoid race conditions. Jared ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setvar SIP_CODEC
[extensions.conf]
exten = 123456,1,SetVar,SIP_CODEC=ulaw
exten = 123456,2,Dial(${TRUNK}/${EXTEN})
The problem is with the SetVar function, the debug shows that the
function is executed, but after that, * sends the media capability to
the phone with g729 as preferred codec.
SIP_CODEC is was supposed to only change the codec of the incoming call,
eg: asterisk responds with ANSWER with ulaw codec ...
But it won't change anything with the 2nd call.
regards
Martin
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Re: [Asterisk-Users] A software FAX modem
On Tuesday 21 October 2003 12:01 pm, Maik Schmitt wrote: We tried to use it witch our AVM Fritz!Card with chan_capi but asterisk always crashes after our fax-machines shows the ID of the soft-fax (12345678). Here's a backtrace: #0 0x417546a4 in TIFFWriteBufferSetup () from /usr/lib/libtiff.so.3 #1 0x417547fd in TIFFFlushData1 () from /usr/lib/libtiff.so.3 As this is a separate project, shouldn't it have it's own mailing list and web site. ie. sourceforge. It's OK to announce it, but if everyone added posts about other software that turned into support and maintenance of said software, then this list is going to become unusable. Regards...Martin PS. Don't shott the messenger -- Gee, Toto, I don't think we are in Kansas anymore. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A software FAX modem
On Tue, 2003-10-21 at 16:22, Steve Underwood wrote: I did say this was a first test release :-) I can't be held responsible for libtiff being empty on your machine, but they other issues are my fault. I have put a new tarball up, which should need nothing more than libtiff for the library to work. All the other libraries you had issues with are only needed by the test programs. ./configure will now only set things up to build the library. ./configure --enable-doc --enable-tests --enable-itutests will set things up to be everything. Looks like I've got a specific Mandrake problem. Is any one else trying MDK 9.0? 9.0 does not have the aclocal-1.6/automake-1.6 combination so Makefile.am fails, I'll have to update them. With 9.2 ./configure works but make barks because tiffiop.h requires tif_dir.h which is nowhere to be found. Has anyone got a copy of that, or know which package it is in? I really want to give this a try because the one thing I haven't got is a working fax. Thanks for your hard work Steve, you've done the hardest bit now it's just to sort out Mandrake, comme d'habitude. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Send to VoiceMail button
I know this is going to sound like a strange question, but here goes: Does anyone know of a SIP softphone that has either a button or a programmable soft-key to send the current call to VoiceMail? I'd appreciate any ideas you might have. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Defragmenting mailboxes
--- Jared Smith [EMAIL PROTECTED] wrote: On Tue, 2003-10-21 at 09:44, Ryan Tucker wrote: Does anyone have a quick and dirty script for defragmenting mailboxes? [snip] Note the gap between 0003 and 0009. This is caused by a somewhat common situation, and it tends to bite us somewhat often. :-) If not, if I get a chance, I'll whip something up. -rt Yes, the gaps in the numbering get really annoying. Unfortunately, it's a little bit risky to go moving files around without stopping asterisk first, or you might just contribute to the problem instead of helping it. (For example, if someone is leaving a message while you are renumbering the files.) I think the best long-term solution is to [ask|beg|pay|coerce|convince] someone to fix the way voicemail messages are numbered to avoid race conditions. There is a C Library function that will return a unique file name. (see man mkstemp) That's the best way to go. It is generally a bad design to encode any information in a file name. Better to simply use the file's date/time stamp to order the messages. defragmenting is a rather poor band aid type fix. Think about the case where the defragmentor is runnig while multiple inbound callers are all leaving voicemail while at the same time the user is listening to his voise mail. Yes this can happen on a popular mailbox. Oh and then there is the case where a voicemail file was e-mailed and now the user wants to find the voicemail file that matchs the one he was e-mailed. No, don't go changing file names. Ideally this kind of stuff would go into a DBMS. That would not only fix any race condition but also allow software other then Asterisk to safely access the data. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM 200 beta build + MOH
I'm using the SNOM 200 latest SIP beta (so that I can have the GSM codec, etc). Everything seems to be working fine, but the music on hold doesn't play when I use the HOLD button on the snom. Any suggestions? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Send to VoiceMail button
At 09:53 AM 10/21/2003, you wrote:
I know this is going to sound like a strange question, but here goes:
Does anyone know of a SIP softphone that has either a button or a
programmable soft-key to send the current call to VoiceMail?
Here is what I use with a SNOM 200...
exten = _2,1,Voicemail2(u${EXTEN:1})
Then, configure any of the SNOM's redirect options (automatic or using key
mapping) to redirect to your extension with a 2 prepended. Works perfectly.
--Ernest
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Re: [Asterisk-Users] Send to VoiceMail button
No I think he means on the phone.. like a softkey to do it.
On Tue, 21 Oct 2003, Ernest W. Lessenger wrote:
At 09:53 AM 10/21/2003, you wrote:
I know this is going to sound like a strange question, but here goes:
Does anyone know of a SIP softphone that has either a button or a
programmable soft-key to send the current call to VoiceMail?
Here is what I use with a SNOM 200...
exten = _2,1,Voicemail2(u${EXTEN:1})
Then, configure any of the SNOM's redirect options (automatic or using key
mapping) to redirect to your extension with a 2 prepended. Works perfectly.
--Ernest
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Re: [Asterisk-Users] Defragmenting mailboxes
There is a C Library function that will return a unique file name. (see man mkstemp) That's the best way to go. It is generally a bad design to encode any information in a file name. Better to simply use the file's date/time stamp to order the messages. I was speaking with tclark on IRC about this this past weekend. What is wrong with using Maildir/ type interfaces for voicemail? Maildir is a very straightforward, scalable and distributable way of storing things like email (and voicemail). Each mailbox has this format: ./ tmp/ cur/ new/ When a new voicemail is created, you mkstemp in tmp/ and create the file. Once it's done, you mv it to /new. When it's listened to or otherwise accessed, it's mv'd to cur where it stays until deletion. So to recap: create and manipulate in tmp/, move to new/ once done. When no longer new, move to cur/ and leave there. No funky locking, totally NFS safe and very fast, since each voicemail is just a file. There's no patents or any kind of software encumberances to this technique, either. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A software FAX modem
As this is a separate project, shouldn't it have it's own mailing list and web site. ie. sourceforge. It's OK to announce it, but if everyone added posts about other software that turned into support and maintenance of said software, then this list is going to become unusable. Regards...Martin PS. Don't shott the messenger Once this is stable it will be part of * and in the digium cvs... so the discussion is valid. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A software FAX modem
Title: RE: [Asterisk-Users] A software FAX modem The same problem with tif_dir.h is on RH9, make fails because of that. Alex Zarubin -Original Message- From: Dave Cotton [mailto:[EMAIL PROTECTED]] Sent: Tuesday, October 21, 2003 11:45 AM To: Asterisk List Subject: Re: [Asterisk-Users] A software FAX modem On Tue, 2003-10-21 at 16:22, Steve Underwood wrote: I did say this was a first test release :-) I can't be held responsible for libtiff being empty on your machine, but they other issues are my fault. I have put a new tarball up, which should need nothing more than libtiff for the library to work. All the other libraries you had issues with are only needed by the test programs. ./configure will now only set things up to build the library. ./configure --enable-doc --enable-tests --enable-itutests will set things up to be everything. Looks like I've got a specific Mandrake problem. Is any one else trying MDK 9.0? 9.0 does not have the aclocal-1.6/automake-1.6 combination so Makefile.am fails, I'll have to update them. With 9.2 ./configure works but make barks because tiffiop.h requires tif_dir.h which is nowhere to be found. Has anyone got a copy of that, or know which package it is in? I really want to give this a try because the one thing I haven't got is a working fax. Thanks for your hard work Steve, you've done the hardest bit now it's just to sort out Mandrake, comme d'habitude. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Defragmenting mailboxes
Jared Smith wrote: I think the best long-term solution is to [ask|beg|pay|coerce|convince] someone to fix the way voicemail messages are numbered to avoid race conditions. Here's a thought: don't use the filenames to determine the order the messages were left, use ctime or mtime on one of the files - then it doesn't matter what the filename is :) -- JustThe.net Internet Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
10 Fix call waiting tone. 9Fix the tftp configs so that I can host my own provisioning server. Or make a command prompt based tool kit, so that I can use Gaps with out writing a http screen scraper. 4 Having the Conference button do something would be cool. John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ 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] A software FAX modem
Its still broken... hrm #0 0x420743da in _int_realloc () from /lib/i686/libc.so.6 #1 0x42073416 in realloc () from /lib/i686/libc.so.6 #2 0x477ea074 in _TIFFrealloc (p=0x477b7d98, s=20) at tif_unix.c:189 #3 0x477b82a7 in t4_rx_start_page () from /usr/lib/asterisk/modules/app_rxfax.so #4 0x477b16d0 in key () from /usr/lib/asterisk/modules/app_rxfax.so #5 0x477b65c1 in v29_rx () from /usr/lib/asterisk/modules/app_rxfax.so #6 0x477b499b in fax_rx_process () from /usr/lib/asterisk/modules/app_rxfax.so #7 0x477b1145 in rxfax_exec (chan=0x816ec10, data=0x47a4125c) at app_rxfax.c:183 #8 0x08063409 in pbx_exec (c=0x816ec10, app=0x81332e0, data=0x47a4125c, newstack=1) at pbx.c:396 #9 0x0806a5f0 in pbx_extension_helper (c=0x47a4125c, context=0x816ed68 default, exten=0x816ee5c s, priority=1, callerid=0x80db628 \X \ , action=135719952) at pbx.c:1151 #10 0x080652fc in ast_pbx_run (c=0x816ec10) at pbx.c:1635 #11 0x41d0eed2 in ss_thread (data=0x816ec10) at chan_zap.c:4386 #12 0x40024941 in pthread_start_thread () from /lib/i686/libpthread.so.0 On Tue, 21 Oct 2003, Dave Cotton wrote: On Tue, 2003-10-21 at 16:22, Steve Underwood wrote: I did say this was a first test release :-) I can't be held responsible for libtiff being empty on your machine, but they other issues are my fault. I have put a new tarball up, which should need nothing more than libtiff for the library to work. All the other libraries you had issues with are only needed by the test programs. ./configure will now only set things up to build the library. ./configure --enable-doc --enable-tests --enable-itutests will set things up to be everything. Looks like I've got a specific Mandrake problem. Is any one else trying MDK 9.0? 9.0 does not have the aclocal-1.6/automake-1.6 combination so Makefile.am fails, I'll have to update them. With 9.2 ./configure works but make barks because tiffiop.h requires tif_dir.h which is nowhere to be found. Has anyone got a copy of that, or know which package it is in? I really want to give this a try because the one thing I haven't got is a working fax. Thanks for your hard work Steve, you've done the hardest bit now it's just to sort out Mandrake, comme d'habitude. -- Dave Cotton [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] Auto-dial from webpage
Interesting... I see that you are not just reordering the lines but
putting the information in different places. Can you give an example of
what the Context would actually look like for this? I tried this with
Context: extensions and that did not work.
elrod
WipeOut wrote:
John Todd wrote:
I want to create a CGI that will allow me to make a call when a user
clicks on a URL in a webpage. I believe I need to create a file in
/var/spool/asterisk/outgoing that defines the number I want to call and
the phone I want to connect it to but I see no way to define the phone
number I want to dial in the /usr/src/asterisk/sample.call file I see
mentioned in other posts. Is it possible to do what I want? Am I even
looking in the right direction?
elrod
--
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Vindicia, Inc.
2755 Campus Drive, Suite 240
San Mateo, California 94403
Email: [EMAIL PROTECTED]
Cell: 650-483-5763
Main: 650-292-2409
Fax: 650-345-1165
Web: http://www.vindicia.com
Mark -
Hello! To create a call to 14109850123 on an analog channel in
group 2 and then connect it to the hypothetical extension 84 (which
would map to 84,1,Dial(SIP/84) ) inside your network, here's the file
you'd create in /var/spool/asterisk/outgoing.call:
#
# Create the call on group 2 dial lines and set up
# some re-try timers
#
Channel: Zap/g2/14109850123
MaxRetries: 2
RetryTime: 60
WaitTime: 30
#
# Assuming that your local extensions are kept in the
# context called [extensions]
#
Context: extensions
Extension: 84
Priority: 1
I would strongly suggest that you create this file elsewhere, and
copy it in after you're done creating it. Asterisk is very
aggressive in grabbing these files, and if you're still creating it
when it grabs the file, you'll get errors, so best to create first
and then copy in to the outgoing directory all at once.
JT
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From your example you would dial the outbound line and once connected
dial the internal extension.. does this not mean that the external
person would hear the phone rining when it was answered??
I do mine the other way around..
Channel: SIP/84
WaitTime: 30
Context: {the context that has your outbound dial string}
Extension: 914109850123 ;9 added to number as if call was being placed
from a phone
Priority: 1
This way the internal extension is dialed first and when answerd the
external number is dialed..
Later..
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Vindicia, Inc.
2755 Campus Drive, Suite 240
San Mateo, California 94403
Email: [EMAIL PROTECTED]
Cell: 650-483-5763
Main: 650-292-2409
Fax: 650-345-1165
Web: http://www.vindicia.com
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Re: [Asterisk-Users] Defragmenting mailboxes
That's very close to my suggestion. It is scalable but only to a point. As soon as you are so big as to require multiple Asterisks servers you will have the same problem as the guys who run large e-mail servers. Te first step would be to NFS mount the mail dir from an NFS server running some kind of RAID. Don't laugh. Around here they have 5,000 voicemail boxes with 25MB limits on each. The current pthreads based locks don't work across mutiple servers so something needs to be done once you move out of the small office environment. The maildir design would work for up to a few thousand users I like DBMS based designs as they make web based interfaces easy to implement and would scale to unlimited size, say to someone like Verizon with a few tens of million of users. --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: There is a C Library function that will return a unique file name. (see man mkstemp) That's the best way to go. It is generally a bad design to encode any information in a file name. Better to simply use the file's date/time stamp to order the messages. I was speaking with tclark on IRC about this this past weekend. What is wrong with using Maildir/ type interfaces for voicemail? Maildir is a very straightforward, scalable and distributable way of storing things like email (and voicemail). Each mailbox has this format: ./ tmp/ cur/ new/ When a new voicemail is created, you mkstemp in tmp/ and create the file. Once it's done, you mv it to /new. When it's listened to or otherwise accessed, it's mv'd to cur where it stays until deletion. So to recap: create and manipulate in tmp/, move to new/ once done. When no longer new, move to cur/ and leave there. No funky locking, totally NFS safe and very fast, since each voicemail is just a file. There's no patents or any kind of software encumberances to this technique, either. Regards, Andrew ___ 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!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Defragmenting mailboxes
The current pthreads based locks don't work across mutiple servers so something needs to be done once you move out of the small office environment. The maildir design would work for up to a few thousand users I like DBMS based designs as they make web based interfaces easy to implement and would scale to unlimited size, say to someone like Verizon with a few tens of million of users. Maybe a backed that stores the mail message via LMTP into an actual mailbox? Servers like Cyrus have solved this problem a long time ago. Only question would be how to play back the message, but you could just store the IMAP message id (always unique). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Defragmenting mailboxes
On Tue, 2003-10-21 at 13:38, Adam Williams wrote: The current pthreads based locks don't work across mutiple servers so something needs to be done once you move out of the small office environment. The maildir design would work for up to a few thousand users I like DBMS based designs as they make web based interfaces easy to implement and would scale to unlimited size, say to someone like Verizon with a few tens of million of users. Maybe a backed that stores the mail message via LMTP into an actual mailbox? Servers like Cyrus have solved this problem a long time ago. Only question would be how to play back the message, but you could just store the IMAP message id (always unique). Once you go this route, you can ignore the local filesystem problem and just mail the file off. Of course this would be interesting in that you could have many users have access to the mailbox. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Tuesday 21 October 2003 10:52, Brian West wrote: We have a 10 and we need it yesterday (as well as many other people who don't even know it). We have a Bug report at GS. The problem is with STUN and changing IP Addresses. It happens like this: 1. Phone does a STUN query and registers fine. 2. If the public IP Address changes sometime later (like on a DSL line that disconnects and connects back), the phone will keep registering with the original IP address, and thus will fail to work properly. It apparently does not attempt further STUN queries for registration purposes. STUN isn't even needed nat=yes is all you need and it just works. We only use Asterisk for PSTN calls. All our subs register in SER, and sure, we could also do the above trick in SER as well, but that would force the RTP stream to pass though our server. We try to avoid it if possible and STUN is a great way to do it. Regards, Andres ___ 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] Weird IAX2 problem
Shaun
Thanks
I finally got it working
You where correct. Removing the callerid from the iax.conf file allowed my
DID callerid to show up on the destination phones. I have no idea why the
callerid field in the iax.conf would overwrite the inbound callerid (I
thought it was for outbound callerid). In fact, I'm not sure why you would
change your inbound caller id anyways.
As for outbound callerid. I tried what you suggested, but it didn't work for
me. I tried something a little different (added the {EXTEN}before the ARG1)
and now it works correctly. Here is my configuration
exten = _1NXXNXX,1,SetCallerID(Lee Goodman(978) XXX-)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}/${ARG1}
I sent a message to Operations at Voicepulse, letting them know this config
works for their service. Also, I think the callerid in the iax.conf file
effecting inbound callerid is a bug. I'm going to enter a bug in the bug
list
Thanks again for all your help
Lee
- Original Message -
From: Shaun Ewing [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 16, 2003 11:56 AM
Subject: Re: [Asterisk-Users] Weird IAX2 problem
When I get a inbound call on my Voicepulse DID, the call hits my
asterisk
server correctly with the correct callerid (the DID phone number
617902). when the call gets passed on to a softphone (X-lite), the
caller id that shows up on the X-lite softphone as Lee , that is the
one
that I have set for outbound Voicepulse calls (callerid=Lee
978xxx)
Any ideas
Take out the callerid= line. Instead, for outgoing calls, set it
explicitly
(shown below).
Also, and I don't know if it's related, No matter what I set my callerid
to
, my outbound calls to Voicepulse always get the default 00
callerid
Try something like:
exten = s,1,SetCallerID(Lee (978) xxx-)
exten = s,2,Dial,IAX2/[EMAIL PROTECTED]/${ARG1}|60|r
I use Macros, hence the ARG1
Seems to work fine (I also use voicepulse for incoming+outgoing).
Thanks
Lee Goodman
-Shaun
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Re: [Asterisk-Users] Defragmenting mailboxes
That's very close to my suggestion. It is scalable but only to a point. As soon as you are so big as to require multiple Asterisks servers you will have the same problem as the guys who run large e-mail servers. Te first step would be to NFS mount the mail dir from an NFS server running some kind of RAID. Don't laugh. Around here they have 5,000 voicemail boxes with 25MB limits on each. I think I'd run CODA or InterMezzo or some other DFS before I hit that point. The current pthreads based locks don't work across mutiple servers so something needs to be done once you move out of the small office environment. The maildir design would work for up to a few thousand users ... it's working for 15000 mail accounts on a busy little dialup ISP... that is over NFS though. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A software FAX modem
Title: RE: [Asterisk-Users] A software FAX modem Found tif_dir.h, make and install look OK. Now it's a coredump: #0 0x4b4ba652 in _TIFFFax3fillruns () from /usr/lib/libtiff.so.3 (gdb) bt #0 0x4b4ba652 in _TIFFFax3fillruns () from /usr/lib/libtiff.so.3 #1 0x4b4074d4 in t4_rx_end_page () from /usr/lib/asterisk/modules/app_rxfax.so #2 0x4b40062e in key () from /usr/lib/asterisk/modules/app_rxfax.so #3 0x4b4055c1 in v29_rx () from /usr/lib/asterisk/modules/app_rxfax.so #4 0x4b40399b in fax_rx_process () from /usr/lib/asterisk/modules/app_rxfax.so #5 0x4b400145 in rxfax_exec (chan=0x81b68a8, data="" at app_rxfax.c:183 #6 0x080633ca in pbx_exec (c=0x81b68a8, app=0x81b1580, data="" newstack=1) at pbx.c:396 #7 0x0806a611 in pbx_extension_helper (c=0x81b68a8, context=0x81b69fc webley_pstn, exten=0x4b8ffb6c /usr/tmp/asteriskfax.tif, priority=1, callerid=0x80d6d58 8478106018, action="" at pbx.c:1151 #8 0x0806528c in ast_pbx_run (c=0x81b68a8) at pbx.c:1635 #9 0x0806acd1 in pbx_thread (data="" at pbx.c:1856 #10 0x4003e9b1 in pthread_start_thread () from /lib/i686/libpthread.so.0 (gdb) Comments are welcome. Thank you. Alex Zarubin -Original Message- From: Dave Cotton [mailto:[EMAIL PROTECTED]] Sent: Tuesday, October 21, 2003 11:45 AM To: Asterisk List Subject: Re: [Asterisk-Users] A software FAX modem On Tue, 2003-10-21 at 16:22, Steve Underwood wrote: I did say this was a first test release :-) I can't be held responsible for libtiff being empty on your machine, but they other issues are my fault. I have put a new tarball up, which should need nothing more than libtiff for the library to work. All the other libraries you had issues with are only needed by the test programs. ./configure will now only set things up to build the library. ./configure --enable-doc --enable-tests --enable-itutests will set things up to be everything. Looks like I've got a specific Mandrake problem. Is any one else trying MDK 9.0? 9.0 does not have the aclocal-1.6/automake-1.6 combination so Makefile.am fails, I'll have to update them. With 9.2 ./configure works but make barks because tiffiop.h requires tif_dir.h which is nowhere to be found. Has anyone got a copy of that, or know which package it is in? I really want to give this a try because the one thing I haven't got is a working fax. Thanks for your hard work Steve, you've done the hardest bit now it's just to sort out Mandrake, comme d'habitude. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
10. Auto answer option on 2nd line appearance. To support paging over the phones. Lee - Original Message - From: John Brown (CV) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 20, 2003 10:38 PM Subject: [Asterisk-Users] Survey: Grandstream improvements. Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ 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] Survey: Grandstream improvements.........
10. Auto answer option on 2nd line appearance. To support paging over the phones. That would be very cool. Voice Call I think it's called on the Meridian system. DND would be nice too (just return busy) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setvar SIP_CODEC
Martin, Thank you for replaying. That's exactly what I am trying to do, but the call never gets answered because is dropped before that due codec incompatibility. Please see what the debug shows with my comments in line. Regards, Luis == == INVITE from the phone with G729 as preferred codec == == Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060 From: User ID sip:[EMAIL PROTECTED];tag=000b5f800a9b010359818116-229fbd39 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Tue, 21 Oct 2003 17:38:25 GMT CSeq: 101 INVITE User-Agent: CSCO/4 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Content-Type: application/sdp Content-Length: 189 Accept: application/sdp v=0 o=Cisco-SIPUA 727 26778 IN IP4 192.168.1.13 s=SIP Call c=IN IP4 192.168.1.13 t=0 0 m=audio 22436 RTP/AVP 18 0 8 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 == == Asterisk asks for authentication == == 13 headers, 9 lines Using latest request as basis request Sending to 192.168.1.13 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found audio format ALAW Found description format G729 Found description format PCMU Found description format PCMA Capabilities: us - 2147483647, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 0, combined - 0 DEBUG[114696]: File chan_sip.c, Line 3854 (check_user): Setting NAT on RTP to 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.13:5060 From: User ID sip:[EMAIL PROTECTED];tag=000b5f800a9b010359818116-229fbd39 To: sip:[EMAIL PROTECTED];tag=as225c5d68 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=2a32fc8f Content-Length: 0 to 192.168.1.13:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060 From: User ID sip:[EMAIL PROTECTED];tag=000b5f800a9b010359818116-229fbd39 To: sip:[EMAIL PROTECTED];tag=as225c5d68 Call-ID: [EMAIL PROTECTED] Date: Tue, 21 Oct 2003 17:38:25 GMT CSeq: 101 ACK Content-Length: 0 8 headers, 0 lines DEBUG[114696]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Found == == Phone sends authentication == == Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060 From: User ID sip:[EMAIL PROTECTED];tag=000b5f800a9b010359818116-229fbd39 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Tue, 21 Oct 2003 17:38:25 GMT CSeq: 102 INVITE User-Agent: CSCO/4 Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=7601,realm=asterisk,uri=sip:192.168.1.111,response=f35280ce287b45e2abdcb832d7244198,nonce=2a32fc8f,algorithm=md5 Expires: 180 Content-Type: application/sdp Content-Length: 189 v=0 o=Cisco-SIPUA 727 26778 IN IP4 192.168.1.13 s=SIP Call c=IN IP4 192.168.1.13 t=0 0 m=audio 22436 RTP/AVP 18 0 8 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 13 headers, 9 lines Using latest request as basis request Sending to 192.168.1.13 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found audio format ALAW Found description format G729 Found description format PCMU Found description format PCMA Capabilities: us - 2147483647, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 0, combined - 0 DEBUG[114696]: File chan_sip.c, Line 3854 (check_user): Setting NAT on RTP to 0 DEBUG[114696]: File chan_sip.c, Line 4904 (handle_request): Check for res for 7601 DEBUG[114696]: File chan_sip.c, Line 973 (find_user): Call from user '7601' is 1 out of 0 Looking for 17862862705 in intern DEBUG[114696]: File chan_sip.c, Line 3307 (build_route): build_route: Contact hop: sip:[EMAIL PROTECTED]:5060list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT): == == Asterisk has authorized the call and sends the Trying to the phone == == SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.13:5060 From: User ID sip:[EMAIL PROTECTED];tag=000b5f800a9b010359818116-229fbd39 To: sip:[EMAIL PROTECTED];tag=as2ae322ec Call-ID: [EMAIL
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 2003-10-21 at 11:36, James Sizemore wrote: 9Fix the tftp configs so that I can host my own provisioning server. Or make a command prompt based tool kit, so that I can use Gaps with out writing a http screen scraper. So I'm not the only one who wrote an http screen scraper to handle configuring a network of phones? :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Mon, 2003-10-20 at 20:38, John Brown (CV) wrote: So please rate your ideas on a scale of 1-10 10 - Fix the TCP/IP stack. The phones don't work with certain switches (i.e. the one at my house), and occasionally do other weird things (although they fixed the MAC address takeover bug, apparently). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Free g.729.1 implementation
1st. - I'm from Poland, we don't have (yet, and hopefully forever) software patents. Is there any free g.729.1 implementation for asterisk? I want to use it for my private use (dialing into inet-PSTN gateway), and I don't want (now) to buy codec, as I don't know if I will be using this service in future (now I just want to test it). Any solutions? Maybe even free-15day-trial of g.729.1 codec? -- Witold Krcicki (adasi) adasi [at] culm.net GPG key: 7AE20871 http://www.culm.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Waiting on SIP phones
On Tue, Oct 21, 2003 at 09:32:44AM +1000, Paul Liew wrote:
Sorry, to repost - but I left a /* comment - here it is again
Paul
--- chan_sip.c.save 2003-10-20 21:51:52.0 +1000
+++ chan_sip.c 2003-10-21 09:26:41.0 +1000
@@ -959,7 +959,9 @@
return 0;
}
switch(event) {
+ /* Incoming and outging affects the inUse counter */
case DEC_IN_USE:
+ case DEC_OUT_USE:
if ( u-inUse 0 ) {
u-inUse--;
} else {
@@ -967,6 +969,7 @@
}
break;
case INC_IN_USE:
+ case INC_OUT_USE:
if (u-incominglimit 0 ) {
if (u-inUse = u-incominglimit) {
ast_log(LOG_ERROR, Call from user '%s'
rejected due to usage limit of %d\n, u-name, u-incominglimit);
@@ -977,6 +980,8 @@
u-inUse++;
ast_log(LOG_DEBUG, Call from user '%s' is %d out of %d\
n, u-name, u-inUse, u-incominglimit);
break;
+ /* Commented out - don't want to limit outgoing */
+ /*
case DEC_OUT_USE:
if ( u-outUse 0 ) {
u-outUse--;
@@ -994,6 +999,7 @@
}
u-outUse++;
break;
+ */
default:
ast_log(LOG_ERROR, find_user(%s,%d) called with no even
t!\n,u-name,event);
}
@@ -1086,6 +1092,12 @@
INVITE, but do set an autodestruct just in ca
se. */
needdestroy = 0;
sip_scheddestroy(p, 15000);
+ /* channel still up - reverse dec of inuse count
er */
+ if ( p-outgoing ) {
+ find_user(p, INC_OUT_USE);
+ } else {
+ find_user(p, INC_IN_USE);
+ }
} else {
char *res;
if (ast-hangupcause ((res = hangup_cause2sip
(ast-hangupcause {
Paul, I'm getting a patch error when I diff to the chan_sip.c that I just got from CVS
this morning. It looks like this morning's version hasn't changed from the version I
had from 9/24/03. Here's the .rej file output:
***
*** 1071,1076
INVITE, but do set an autodestruct just in case. */
needdestroy = 0;
sip_scheddestroy(p, 15000);
} else {
char *res;
if (ast-hangupcause ((res =
hangup_cause2sip(ast-hangupcause {
--- 1080,1091
INVITE, but do set an autodestruct just in case. */
needdestroy = 0;
sip_scheddestroy(p, 15000);
+ /* channel still up - reverse dec of inuse counter */
+ if ( p-outgoing ) {
+ find_user(p, INC_OUT_USE);
+ } else {
+ find_user(p, INC_IN_USE);
+ }
} else {
char *res;
if (ast-hangupcause ((res =
hangup_cause2sip(ast-hangupcause {
Here's what I find in the source around those lines:
needdestroy = 1;
/* Start the process if it's not already started */
if (!p-alreadygone strlen(p-initreq.data)) {
if (needcancel) {
if (p-outgoing) {
transmit_request_with_auth(p, CANCEL, p-ocseq, 1);
/* Actually don't destroy us yet, wait for the 487 on
our original
INVITE, but do set an autodestruct just in case. */
needdestroy = 0;
sip_scheddestroy(p, 15000);
} else
transmit_response_reliable(p, 403 Forbidden,
p-initreq);
} else {
if (!p-pendinginvite) {
/* Send a hangup */
transmit_request_with_auth(p, BYE, 0, 1);
} else {
/* Note we will need a BYE when this all settles out
but we can't send one while
Re: [Asterisk-Users] Free g.729.1 implementation
Witold Krecicki wrote: 1st. - I'm from Poland, we don't have (yet, and hopefully forever) software patents. Is there any free g.729.1 implementation for asterisk? I want to use it for my private use (dialing into inet-PSTN gateway), and I don't want (now) to buy codec, as I don't know if I will be using this service in future (now I just want to test it). Any solutions? Maybe even free-15day-trial of g.729.1 codec? Won't happen. Just spend the $10 and support Asterisk. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Agreed, don't drive up my shipping cost. light is good. Tilghman Lesher wrote: I'd have to respectfully disagree. If this is really a problem I'd suggest taking advantage of the mounting bracket on the bottom and either attach the phone to the desk or attach a sheet of lead. -Tilghman ___ 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] #include in config /New subject/
Steve Creel wrote: You'll want to #include it. This leaves the burden of the [general] and any static configs on sip.conf but allows the script to blindly write out from the database to sip_additional.conf in sip.conf: #include sip_additional.conf Eureka! ...is this #include construct a general command for all config files? I must have missed it - where is it documented? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free g.729.1 implementation/Open G.729
go to the Vovida site http://www.vovida.org/ and checkout the Open G.729(A) Initiative There is an open source g.729 there --- Witold Krecicki [EMAIL PROTECTED] wrote: 1st. - I'm from Poland, we don't have (yet, and hopefully forever) software patents. Is there any free g.729.1 implementation for asterisk? I want to use it for my private use (dialing into inet-PSTN gateway), and I don't want (now) to buy codec, as I don't know if I will be using this service in future (now I just want to test it). Any solutions? Maybe even free-15day-trial of g.729.1 codec? -- Witold Krêcicki (adasi) adasi [at] culm.net GPG key: 7AE20871 http://www.culm.net ___ 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!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
