Re: [Asterisk-Users] chan_oss.c:461: error: too many arguments to function `ast_queue_frame'
Alessio Focardi wrote: I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/07/04-11:28:50\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC-c -o chan_oss.o chan_oss.c chan_oss.c: In function `oss_call': chan_oss.c:461: error: too many arguments to function `ast_queue_frame' chan_oss.c:467: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `oss_new': chan_oss.c:712: warning: assignment from incompatible pointer type chan_oss.c: In function `console_answer': chan_oss.c:809: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `console_sendtext': chan_oss.c:841: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `console_hangup': chan_oss.c:861: error: too many arguments to function `ast_queue_hangup' chan_oss.c: In function `console_dial': chan_oss.c:883: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `console_transfer': chan_oss.c:935: error: too many arguments to function `ast_async_goto' make[1]: *** [chan_oss.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 It appears that the final argument to all these functions (normally a 0 or 1) has been dropped, but it hasn't been fixed in chan_oss.c or chan_alsa.c. If you happen to have already compiled asterisk before and aren't doing a clean recompile then it appears that the problem isn't spotted and recompiled (poor dependency checking in the Makefile?) The easy fix is just to drop the final arguments for all these functions and then to kick off the compile again. Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
Mark Phillips wrote: What codecs are you using? You should be able to get quite a few speex channels or even a load of 729 or gsm channels down your 256K. Mark You always have to remember that a UK ADSL line has a contention ratio of 20:1 if you have business ADSL or 50:1 if you have the consumer option. This means that in the worst event you could find out that your 256K on a business line is actually 12Kbit/s and the consumer option is only around 1Kbit/s. These are worst case scenario, but it would mean event one GSM call was too big to get down the pipe. In reality I suspect you would probably never see more than 4-5 people competing with you, but even then it is only around 50-64Kbit/s. I often find I get pretty much the full pipe, but you can't really plan on that and it would be bad to run too many calls over the line at a time. Typically we find that 5 is about the reasonable limit. Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference and transfer
Joel Maslak wrote: My understanding is that the purpose of the button is to look pretty unless you have the higher-end budget-tone (102?) where it then does 3 way calling. Doesn't work on my BT102 phones. Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream transfer into outer space
Jim Rosenberg wrote: The Grandstream BudgeTone 101 phone has a Transfer button. This appears to be a blind transfer: once you've dialed the extension to which you want to transfer, the phone tries to do this and then dumps you out. You also get similar trouble if people press the transfer button, but forget to press 'send' at the end (certainly I find the calls are not transferred until 'send' is pressed, or maybe they just take a little while). The big problem is that with the sometimes unreliable budgetone keyboard you tend to lose the call due to mistyped numbers. To avoid this you could create some catch-all extensions that automatically forward calls back to reception (or your voicemail system). Something like: exten = _[0-9],1,Goto(s,1) exten = _[0-9][0-9],1,Goto(s,1) Unfortunately this then prevents you from using # in such a way that it warns you if you type an invalid extension. This sort of system could be further improved. You could set an extension number variable on incoming calls to record the extension which answered the call. Then when a call is transferred to an otherwise invalid extension you wait a few seconds (give them time to put the phone down) and then call the extension number given in the variable. I leave this as an exercise to the reader ;-) Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_sql_postgres doesn't clean up
I am currently looking at using the app_sql_postgres.c stuff, but it almost immediately occurred to me that there is no way of guaranteeing that the whole of the extension code will be executed. As a result I also can't be sure that the postgresql connections will actually be closed. I realise that if I clump together all the PGSQL stuff at the start it should hopefully be able to execute quickly before the user hangs up, but this isn't exactly reliable. Does anybody have any suggestions on how to avoid this problem? Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where to download app_pgsql
I would like to play around with app_pgsql, but where I can download it from seems to be a rather well kept secret. If anybody could tell me where I can download it from then I would greatly appreciate it. Many thanks, Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to download app_pgsql
Steven Critchfield wrote: On Wed, 2004-02-18 at 11:55, Michael T Farnworth wrote: I would like to play around with app_pgsql, but where I can download it from seems to be a rather well kept secret. If anybody could tell me where I can download it from then I would greatly appreciate it. Use the source, Luke. /usr/src/asterisk/apps/app_sql_postgres.c Many thanks. I thought it was called app_pgsql.c so I was never likely to find it :-) I blame that person who wrote in the mailing list that they had used app_pgsql.c to produce app_mysql.c Confusion is King. Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Jitter Buffer Configuration (typo in iax.conf)
I had noticed that the jitterbuffer settings under Asterisk didn't seem to work very well, then I noticed that there was a typo in my iax.conf file where I had: maxexccessbuffer=750 which should have been maxexcessbuffer=750 I have just realised that I didn't make this typo, it is actually a typo in the sample iax.conf file which is provided with Asterisk. People might want to take a look at their own settings and check if you have the same problem! Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reversing a Firmware Upgrade
My Grandstream phone seems quite happy to accept a new firmware, but having tried the latest beta firmware, which I am unhappy with I want to update with an older version. How do I do this? Thanks, Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best SIP PHones to buy ?
On Sat, 20 Dec 2003, Carlos Arnt wrote: I wanna buy a Sip Phone, but what is the best and cheap one ? I see alot of messages about, grandstream , snow etc etc. We have bought around 30 Grandstream phones, both BT101 and BT102. In general the phone is reasonable, but it does have limitations. Notable issues for users tend to be the lack of any sort of consultative transfer or easy access to conference calling. Also its 'call waiting' facility is a rather annoying and loud normal ringing noise, rather than the usual 'beep beep' that people are probably used to and you can't disable the call waiting feature. Entering numbers also has problems as if you dial too quickly you tend to lose digits, even if you heard the tones and saw them appear on the display. We just bought a Snom200 which is a lot more expensive, but you do get what you pay for. It has a better display which copes with caller name as well as caller number, has consultative transfer, conference calling, no problems with losing digits when dialling and a normal 'call waiting' beep. It also has a host of other features, but we haven't used it long enough to find any problems as yet. I don't know what the Snom105 is like, but if it is anywhere near as good as the Snom200 it may be a good alternative to the BT101/2. Can him be used behind a nat system etc ? Both the Snom200 and BT101/2 have settings for use behind NAT. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI Calls Don't Bridge
I had a working configuration whereby an incoming call on an ISDN line would be sent out on the second ISDN line, but since I updated to the latest version of Asterisk I get this error message: WARNING[311315]: File res_parking.c, Line 226 (ast_bridge_call): Bridge failed on channels CAPI[contr1/s]/0 and CAPI[contr1/01624619052]/1 The message comes up as soon as the outgoing call is answered and the call is lost. I have switched to sending the outgoing call using an IAX connection and that does not have the problem. I am also getting this error message (just before the other one), which I don't recall seeing previously: WARNING[311315]: File channel.c, Line 1296 (do_senddigit): Unable to handle DTMF tone 'f' for 'CAPI[contr1/s]/0' Thanks, Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Inuse Count Wrong
I am currently using a copy of Asterisk checked out as the code of 10 days ago from Asterisk and the: sip show inuse reports that I have 3 incoming connections to one of the Grandstream phones, even though that isn't the case. I believe I have tracked the problem down to the following error message, which also (conveniently) showed up 3 times: -- Got SIP response 481 back from 192.168.252.101 Incidentally I am using the code from 10 days ago because as explained earlier today the CAPI support with the current Asterisk is unable to do a native bridge between two CAPI calls (instead it just drops both calls). Incidentally am I sending this report to the right mailing list, or is there a better one? Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Translation time
When you look at 'show translation' you see: Translation times between formats (in milliseconds) but is this the number of milliseconds required to convert 1 packet of data, or the amount of time required to translate 1 second of data? I am assuming it is the time for 1 second. Thanks, Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream Budgetone Phone DHCP General Observations
On Fri, 5 Dec 2003, Nicolas Bougues wrote: On a slightly different topic : does somebody know of a NAT-friendly (as Grandstream means it) tftpd server ? It seems theirs replies from port 69, which is the only thing their phones will accept. [ If anybody wants it, I can send the 1.0.4.17 firmware by email ]. I am slightly puzzled why the 1.0.4.17 firmware isn't the version that Grandstream offers through their tftp if it is the latest version. I just noticed that my new Grandstream phones have 1.0.4.17 whilst the older ones have 1.0.3.81. Can anybody shed any light on the reason? Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CTI/TAPI
I think we need to look carefully at the situation here. We all know that training on complex specialist products can often come in at between 2-4,000USD per week per individual (certainly this is my commercial experience). However normally when I go on a training course there are around 10 other people in the class at least, which amounts to 20-40,000USD for the week of training. It may well be that the company offering the training could use their trainer for instructing a group of 10 people instead, in which case offering the training for less than 20,000USD would result in a loss of revenue. The figure may sound ridiculous, but it might just reflect commercial reality. Mtf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stutter dialtone but no messages
On Sat, 22 Nov 2003, marrandy wrote: Now, I am getting the stutter/dialtone, that tells me I have a message, But when I check both the new and old messages on each of the 3 mailbox accounts, there are no messages. I had this problem the other day. I think it was caused by me trying to pick up a message whilst it was still being recorded. The incoming call I was listening to was cut off, and it also left behind a file in the INBOX. With that file present it continued to stutter the dialtone. After I deleted that one file it stopped. The file just had one line which I think was the time length of the message. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
On Thu, 20 Nov 2003, marrandy wrote: On Monday 17 November 2003 10:31 pm, Brian West wrote: Show us your sip.conf entries.. and i'm sure I can point out the error. bkw Well, I've tried over 20 different settings, from examples in the archives etc. This is the last one I tried, for what it's worth. I had strange registration problems with my Grandstream SIP phones which vanished when I changed from specifying the host in sip.conf to host=dynamic defaultip=192.168.254.160 Not sure if that is at all helpful, but it was rather peculiar. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] which TDM to use? DID line from telco with no dial tone and no voltage
Interesting thought, with these DDI lines a UK based company could easily get a good number of incoming analogue lines into an Asterisk system because teh FXS cards have far more ports than the FXO ones. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone International Calls
Do you know why there are two different possible contexts? Of course it would seem a little strange to put somebody outside the US into the NANPA context rather than the WORLD one ... Michael On Mon, 27 Oct 2003 [EMAIL PROTECTED] wrote: TOP POSTING MADNESS continues... you need to be part of the WORLD context, and not just NANPA, otherwise 011+COUNTRY+AREA+NUMBER works as my numerous jerjer bills will testify -wasim On Sun, 26 Oct 2003, Michael T Farnworth wrote: Does anybody know how to do an international call using NuFone. I realise this isn't really the place to ask, but NuFone appears to be closed for the weekend and would like to have a try at this before tomorrow. I assumed it would be '011' for an international line followed by country code but that doesn't seem to work. and I still trimmed the bush... ___ 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] NuFone International Calls
Does anybody know how to do an international call using NuFone. I realise this isn't really the place to ask, but NuFone appears to be closed for the weekend and would like to have a try at this before tomorrow. I assumed it would be '011' for an international line followed by country code but that doesn't seem to work. I am getting: -- Executing Dial(SIP/phone1-adc5, IAX2/[EMAIL PROTECTED]/011441942XX) in new stack -- Called [EMAIL PROTECTED]/011441942XX WARNING[131081]: File chan_iax2.c, Line 4160 (socket_read): Call rejected by 65.127.126.42: No such context/extension -- Hungup 'IAX2[NuFone]/2' == No one is available to answer at this time -- Executing Congestion(SIP/phone1-adc5, ) in new stack I have tried dropping the 011 and jumping straight to the country code but that doesn't work either. Does anybody have any suggestions? Thanks, Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone International Calls
On Mon, 27 Oct 2003 [EMAIL PROTECTED] wrote: On Sun, 26 Oct 2003, Michael T Farnworth wrote: Do you know why there are two different possible contexts? Of course it would seem a little strange to put somebody outside the US into the NANPA context rather than the WORLD one ... its not where you ARE, its where you're calling... NANPA gives you access to the contigous-US-48, whilst the WORLD pretty much leaves you at your own peril... (ofcourse, none of this is official NuFone, best let jerjer advise you accordingly) I really meant it is a little strange to put somebody who lives outside the US in a context which means they can only call US numbers, because they will almost certainly need to make calls to non-US numbers. I do want to make some US calls, but I did also request, and receive, a copy of the rates for all countries. If nothing else that was a very strong indicator that international calls were on my mind. Even the NANPA context leaves me at my own peril, but I guess it would take somebody longer to use up my money. Of course I would be able to watch out for this if the Subscriber Management System link didn't give me the message 'nothing to see here'. I don't suppose you know where account details are obtained from? By the way, what is jerjer, some sort of play on the name Jeremy McNamara, or his email address jj? Thanks, Mtf - wasim -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
On Sat, 25 Oct 2003 [EMAIL PROTECTED] wrote: Yes I do have gzip installed on that box. Any other ideas? You are looking for the the libz stuff, which if you use RedHat is a part of zlib-devel-1.1.3-25.7 (or whatever the right number is for your distribution). You should be able to see it in somewhere like /usr/lib/libz.a or /usr/lib/libz.so.1.1.3 Michael On Sat, 25 Oct 2003, WipeOut wrote: [EMAIL PROTECTED] wrote: Yes I do have the mysql and mysql-devel packages installed. With my very limited knowledge of C/C++ here's what seems to be the culpret line right before the error: /usr/bin/ld: cannot find -lz Any suggestions here? AJ I think thats gzip.. Have you got gzip installed? ___ 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] CVS update
On Sat, 25 Oct 2003, Rich Adamson wrote: make upgrade -- it connects to CVS and shows me all the new files it downloaded make clean ; make install I just tried the above, and absolutely nothing was updated. I actually checked *.c files for dates/times, and all were from previous cvs. Anything special one needs to do leading up to issuing the above? It is possible you have a sticky tag or date set on your CVS checkout. Try taking a look at the output of: cvs status Makefile Or any other file. To get rid of the sticky tag you would want to do: cvs update -d -A Michael ___ 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] Festival on RH9?
On Thu, 23 Oct 2003, Rich Adamson wrote: Gus, I think I might see the issue. In /etc/asterisk/festival_server.log, I see: Load server start ./festival_server.scm festival port=1314 wrapper Thu Oct 23 20:08:37 CDT 2003 : USING DEFAULT CONFIGURATION wrapper Thu Oct 23 20:08:37 CDT 2003 : waiting serverThu Oct 23 20:08:38 2003 : Festival server started on port 1314 client(1) Thu Oct 23 20:09:08 2003 : rejected from phoenix.routers.com not in ac cess list client(2) Thu Oct 23 20:32:47 2003 : rejected from phoenix.routers.com not in ac cess list Look in /etc/hosts.allow and /etc/hosts.deny. Try the command: man 5 hosts_access to get up the manual page on this topic. You probably need something in /etc/hosts.allow which reads: festival: phoenix.routers.com Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI questions..
On Fri, 24 Oct 2003, Jared Smith wrote: On Fri, 2003-10-24 at 09:04, Steven Critchfield wrote: You can use php, but php is just perl lite. if you do anything more than a trivial app, you will want to be in something more than php. Take for instance all the modules you have in CPAN that can deal with audio, databases and any number of odd things you need to have. I don't mean to start a flame war, but it's opinions like this that drive me nuts... (And just to be fair Steven, I *almost always* agree with your opinions posted to this list. This one just happens to get on my nerves.) Just because you *obviously* prefer Perl, doesn't mean you have to belittle PHP. WipeOot *obviously* wants to use PHP. So (in my humble opinion) we ought to either help him figure out how to do it in PHP, or keep our mouths shut. One of the great things about PHP is that you can just get one download, compile and install it and it has everything you will probably want for web use built in. No need to mess around downloading lots of CPAN things. Okay there might not be as many libraries available, but for web stuff it keeps life simple. Personally I write in either Perl or PHP depending on the job. Programs I run from the command line are written in Perl and web things in PHP. However I feel both products have things to recommend them. In PHP functions and classes are defined in a more conventional way, none of this pass a stack and pop things off to get your function arguments. Perl feels like it was more carefully thought out. Everything is done in relatively generic and flexible ways. Database support is just fine in PHP but audio would admittedly be less supported. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is the X100P a WinModem?
Linux benefits from the fact that it runs on very common hardware and you get commodity prices for most parts of an Asterisk system, and the software is free. Quite honestly the X100P cards cost peanuts for what is a effectively a specialist item. If they cost $10 it wouldn't be financially viable to produce because the market is too small. 10 years ago ethernet cards were more expensive and it was only massive takeup that reduced them to their current prices. If there was a big market you can be sure that somebody would start to produce cheap cards. If somebody out there wants to develop their own hardware/drivers then I am sure nobody will stand in their way. Personally I will stick with the hardware and drivers produced by people who really understand Asterisk, and be grateful for them. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soundcard Problem
On Wed, 22 Oct 2003, Steven Critchfield wrote: Judging by your behavior with this post, I'm betting you are either a RH or Mandrake user, and may have ALSA running with ARTSD also. So you may be in for a little bit of work. Somebody got out of bed on the wrong side this morning, or was this a deliberate attempt to invoke some sort of flame war? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 22 Oct 2003, John Brown (CV) wrote: http == hyper text transport protocol tftp == trivial FILE trasfer protocol Based on this definition we could suggest that the web should only consist of a few html files as a jpeg clearly isn't hypertext. I suspect the reason HTTP was proposed is that almost everybody who runs a network will know about it, and we must also remember that almost every Unix OS is likely to come with a pre-installed and configured webserver, on the other hand tftp is almost always going to be disabled and need configuring. HTTP is trivial to implement on the client side, you simply send a line like: GET /someconfig.txt HTTP/1.0 You then read the input until you get to a blank line, then the file follows that. It isn't hard to write a simple http daemon if people are looking for a small footprint, in fact I am sure such things do exist. You could even write a http daemon using tcp_wrappers and a few lines of shell script if hard pressed. I suspect tftp probably has a simple protocol too. Maybe support could be added for http as well as tftp. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 22 Oct 2003, Michael T Farnworth wrote: I suspect tftp probably has a simple protocol too. Maybe support could be added for http as well as tftp. I take this back, as a protocol tftp is hideously complex compared to http and would take a lot more code. However tftp is based on tcp rather than udp so it requires less complex networking support. Amusingly though SIP has a lot in common with HTTP, so maybe half the work is done already, so much so that in one of the SIP RFC's they even go so far as to say ... Except for the above difference in character sets, much of the message syntax is and header fields are identical to HTTP/1.1; rather than repeating the syntax and semantics here we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1 specification (RFC 2068 [11]). However they do also point out that: Unlike HTTP, SIP MAY use UDP. In fact I believe a SIP client doesn't have to support TCP, but fortunately I believe the Grandstream does. Feel free to point out errors in this brief bit of research. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 22 Oct 2003, Michael T Farnworth wrote: However tftp is based on tcp rather than udp so it requires less complex networking support. Replying to own email here, which is bad I am told, but I did make a mistake, I meant to say tftp is based on udp rather than tcp. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On 22 Oct 2003, Josh Howlett wrote: On Wed, 2003-10-22 at 15:38, Michael T Farnworth wrote: On Wed, 22 Oct 2003, Michael T Farnworth wrote: I take this back, as a protocol tftp is hideously complex compared to http and would take a lot more code. RFC 1350 (tftp v2): 11 pages RFC 2616 (http/1.1) : 114 pages There might be lots more options for http (if you want to use them), but using http in its simplest form is easy. You can do a http request using telnet and typing a single line. I doubt that is possible with tftp (even if there was a udp equivalent of telnet). Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
It just struck me that the easiest improvement would be to drop the name BudgeTone, because 'Budget' tends to imply cheap, which in turn suggests poor quality. The phones need a name which implies 'High Quality'. The current name resulted in my wife bursting into laughter and saying 'I can't believe they called it that.' I suspect clients are more likely to say 'perhaps we need the professional product if we are going to risk our business on it.' 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 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 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] 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] 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
Re: [Asterisk-Users] Odd ringing conditions
On Wed, 15 Oct 2003, Robert Boardman wrote: 1) When an incoming call is detected by asterisk it takes 2 or three rings before the internal phone ring does anyone know how I can fix this? Callerid doesn't work in the UK because BT puts the callerid before the first ring, unlike the US where it after the first ring. If you are using an analogue card it is probably waiting to hear that callerid. Disable the callerid in /etc/asterisk/zapata.conf Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I give up!!
On Thu, 16 Oct 2003, Dave Alan Caruana wrote: no consultative transfer. The closest I got was to park the call, call the other party, tell him a voce which line the call is parked on and then get him to pick up the call. This is, in my opinion, a very basic feature that is missing on asterisk. The park/ pick up sequence proved too difficult for the clients' secretaries to grasp. I agree entirely, this is a particularly awful part of using the GrandStream BudgeTone phones, but the general view seems to be that you need to buy an expensive phone for the reception and then things should be okay (Snom200 was suggested). Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announced Call Transfer
On Wed, 15 Oct 2003, WipeOut wrote: [EMAIL PROTECTED] wrote: I know this topic has been done before but I cant find an answer for it anywhere. I have Grandstream phones running with my Asterisk server. Is there anyway to do announced transfers with these phones? i.e. A is talking to B, A then presses a key to initiate transfer, calls C, tells C that B is on the phone. Then A hangs up and C is connected to B. No, its can't be done with the GS phones, as a work around you could put B on hold then call C tell them about B then go back to B and blind transfer to C.. Which telephones support this requirement and is it possible to have a more expensive phone in reception but leave the other people on the cheap Grandstream phones? Couldn't this problem be solved with an asterisk upgrade? Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream ringer
On Wed, 15 Oct 2003, rnc Info Lists wrote: One option I would definatly like is the ability to turn off the ringer. Since my testing ususally happens after my wife goes to bed it would help NOT to have the audible ring but only the visual indication! Better still I would like volume control over the ringer as the default tends to be rather loud and annoying to other people in the same room. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announced Call Transfer
On Wed, 15 Oct 2003, WipeOut wrote: Michael T Farnworth wrote: more expensive phone in reception but leave the other people on the cheap Grandstream phones? Yes, I have found the Snom200 does consultative transfers well.. Couldn't this problem be solved with an asterisk upgrade? No, Its an issue that is handled on the phone.. Perhaps I am confused, but I tend to believe that Asterisk sits in the middle of all these calls. So when I press the # key for transfer it could actually put the incoming call, allow me to dial and then speak to a person and then when I press # again redirect that incoming call, or maybe I could press * instead if I wasn't going to put the person through. I suppose this would only be possible where the SIP phones weren't doing a native bridge between themselves though. However perhaps an expensive phone in reception is just the easier option. Michael 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] Announced Call Transfer
On Wed, 15 Oct 2003, Rich Adamson wrote: Michael T Farnworth wrote: more expensive phone in reception but leave the other people on the cheap Grandstream phones? Yes, I have found the Snom200 does consultative transfers well.. Couldn't this problem be solved with an asterisk upgrade? No, Its an issue that is handled on the phone.. Perhaps I am confused, but I tend to believe that Asterisk sits in the middle of all these calls. So when I press the # key for transfer it In many cases, that's a bad assumption, but it depends on your config. As I said in the part of my original email which has been cut out: I suppose this would only be possible where the SIP phones weren't doing a native bridge between themselves though. Unless you've purposefully configured something different, asterisk is not in the middle. Once a call is established, the communications (packet flows) happen directly between the two sip phones and does not pass through asterisk. So the problem becomes an issue of the phone itself. What has the phone been programmed to do when any key has been pressed (regardless of whether its the # key or something else)? Certain parts of the operation must go via asterisk though even when there is a native bridge, because when I press the # key asterisk starts to play music to the other telephone and announces to the other phone that it is about to do a transfer. My question is whether more could be done at that point to have an attended transfer. The discussion suggests the Grandstream phones have not been programmed to handle transfers. (I don't have one therefore I don't have a clue as to whether that happens to be the result of the vendor, or the person that has implemented the phone doesn't have the knowledge or documentation to do it.) They have a transfer button and can do a blind transfer. I am really looking for a way to convince Asterisk to do the following in a much easier way: A calls B B parks the calls from A B puts down the phone and picks it up again and calls C C says he/she will take the call B tells C where the call is parked C puts down the phone and picks up the phone and dials the parked call or A calls B B parks teh call from A B puts down the phone and picks it up again and calls C C says he/she will not take the call B puts down the phone and picks up the phone and dials the parked call Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXTEL - Problem Configuration.
Is your problem related to the settings under iax.conf? I have commented out the whole of [iaxtel2] and left in [iaxtel]. I have a problem with my setup in that I have got it to register by putting the register command in the iax.conf, but when I call my own number I just get silence. Output on console is: -- Executing Dial(SIP/phone1-b4ee, IAX/farnwomt:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Calling using options 'exten=17008451426;callerid=192.168.254.160;language=en;context=iaxtel;username=farnwomt;formats=2;capability=65283;version=1;adsicpe=2' -- Called farnwomt:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 12.37.165.130 (format GSM) -- Format for call is GSM Any guesses why this happens? Thanks, Michael On Tue, 14 Oct 2003, Ariel Batista wrote: Ok folks I have another question. So far I have gotten my IAXTEL number and I have been able to make calls from my asterisk system to any IAXTEL number and even to FWD numbers. I also got FWD to work and I now can get calls to my main system. It's great when these things work. But when I call my own IAXTEL number 17005441100 all I get is a message saying the user is un registered or un available. I set the register = loginName:[EMAIL PROTECTED] in the iax.conf. So how do I configure the extensions.conf to send the call to my extension! Here is where I am lost! ___ 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] IAXTEL - Problem Configuration.
Somebody keeps saying it is bad form to respond to one's own postings, but I am going to do it here ... Further experimentation and I discovered that this change to iax.conf make the problem go away: ; ; Trust Caller*ID Coming from iaxtel.com ; ;[iaxtel] ;type=friend ;context=default ;auth=rsa ;inkeys=iaxtel ;[iaxtel2] ; ; Backwards compatible entry for IAXtel pre-RSA ; type=friend context=default deny=0.0.0.0/0.0.0.0 permit=12.37.165.130/255.255.255.255 Michael On Tue, 14 Oct 2003, Michael T Farnworth wrote: Is your problem related to the settings under iax.conf? I have commented out the whole of [iaxtel2] and left in [iaxtel]. I have a problem with my setup in that I have got it to register by putting the register command in the iax.conf, but when I call my own number I just get silence. Output on console is: -- Executing Dial(SIP/phone1-b4ee, IAX/farnwomt:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Calling using options 'exten=17008451426;callerid=192.168.254.160;language=en;context=iaxtel;username=farnwomt;formats=2;capability=65283;version=1;adsicpe=2' -- Called farnwomt:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 12.37.165.130 (format GSM) -- Format for call is GSM Any guesses why this happens? Thanks, Michael On Tue, 14 Oct 2003, Ariel Batista wrote: Ok folks I have another question. So far I have gotten my IAXTEL number and I have been able to make calls from my asterisk system to any IAXTEL number and even to FWD numbers. I also got FWD to work and I now can get calls to my main system. It's great when these things work. But when I call my own IAXTEL number 17005441100 all I get is a message saying the user is un registered or un available. I set the register = loginName:[EMAIL PROTECTED] in the iax.conf. So how do I configure the extensions.conf to send the call to my extension! Here is where I am lost! ___ 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] IAX and Jitter problem
Thought I would just mention that I have a Pentium 150 with 64MB of RAM, asterisk installed, 2 Budgetone 102's and an X100P. No problem with jitter here or anything like that. I don't use mp3 music on hold because I doubt the hardware would cope particularly well. Has anybody got Asterisk running on anything lower spec than this? Michael On Tue, 7 Oct 2003 [EMAIL PROTECTED] wrote: Hello, I've been playing around with * for quite a while now, and have run into a problem that I just cannot seem to figure out. When using * and any IAX client (I have tested with GnoPhone and both clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the connection. What I'm running is a P3-1Ghz machine with 512mb ram for a server. The other end has been various machines (all connected via 100mb switch) ranging from a AMD K6-2 350 running Win98 to another P3-1Ghz running RH Linux 9.0 and GnoPhone. I've tried changing the jitterbuffer settings in iax.conf (including turning it off as I've seen some recommendations on the archives) and I've even tried rebuilding zaptel with the various jitter control switches. At this point I have extension 8500 setup to take me to voicemailmain. When I connect (IAX only - I do not have any Digium cards in the server at all) I can generaly not tell what is being said at all. I've used sox and a player and know that the .gsm files are okay. Anybody have any suggestions of what to try? So far this has been something I've been playing with before I attempt to put it in a production system, but so far am not having a whole lot of luck. I've not been able to try SIP as of yet, as I've not found a softclient and the application I will be using * for would require this. Thanks, Mike Atkinson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else use Audacity for prompts?
I have had been recording my gsm files by getting through to the Asterisk answering service using a GrandStream BudgeTone 102 phone. I then copy the file which is stored in voicemail and use sox to increase the volume. Results are okay but nothing to write home about particularly (or maybe that is just my lack of a good telephone voice). Michael On Mon, 6 Oct 2003, Brian Capouch wrote: Shaun Ewing wrote: I know that some of this comes with the territory, but I wonder if there is anyone out there who does this ( wav - gsm) routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting playback files. What are you using to convert the wav files to gsm? I've been using 'sox' under Linux and have had no quality issues whatsoever. An example line to convert: sox file.wav -r 8000 -c 1 file.gsm That's the exact command I'm using, but it sure does sound crappy to my ears. Perhaps that's the best I can do w/gsm, and of course I expect once the sound is sent through the PSTN most of the highs and bottom are gone anyways. I was just hoping there was something I could do to make the resulting files a bit clearer. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream 102
Typically you run a cable into the phone, then a cable out of the phone into the computer, it appears to just be a bridge. It also works fine to run the cable into the phone and then a cable into another phone and then into a computer. Michael On Sun, 5 Oct 2003, Nicolas Gudino wrote: Sorry about this off-topic question... I want to know if the second ethernet port on the Grandstream 102 phone works as a bridge to connect from there to a PC. Do I need two ethernet jacks to connect a phone and a PC, or this phone let me connect both with only one? Thanks in advance! Nicolas Gudino ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P BT
Sorry, I was confused, I also have an X100P, I should read the subject. Michael On Tue, 30 Sep 2003, Michael T Farnworth wrote: On Tue, 30 Sep 2003, Matthew J Keay wrote: Hi, I've found this thread in the archives and am currently planning to connect my digium kit to a BT landline and also connect a normal UK phone respectively. I've seen a couple of people asking about it but not the answers. My question is - do I need a modtap (listed at maplin.co.uk as BT to RJ45 adapters) full master with line protection, pabx master without line protection or a pabx slave socket. --or-- can I just use a straight-thru rj11 cable for the handset and a straight-thru rj11bt modem cable for the line? I used a simple converter for BT to RJ45 that came with an old modem, it seems to work fine. I tried plugging in a handset cable and it didn't work, in fact phones elsewhere in the house started ringing in a manic way. Michael Matthew ___ 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] Google newsgroup or Forum setup.
On Tue, 30 Sep 2003, Roderick Montgomery wrote: According to Troy Settle: Why do they do that? Quite possibly because they, like myself, hate having to scroll through pages and pages of quotes to get to the reply, which isn't always clear where it might start. Do what? Overtrimming can also have problems of course ... Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Core dumps from Asterisk
Got 3 core dumps from asterisk in a very short period of time, not sure what was going on, but I am posting these in case anybody wants to see them, if you want to see more info I can provide it on request. I am using the asterisk 0.5.0 download. (gdb) bt #0 ast_translator_free_path (p=0x10) at translate.c:83 #1 0x0805b2b9 in ast_set_read_format (chan=0x8109b90, fmts=135250888) at channel.c:1482 #2 0x40575b70 in socket_read (id=0x80e98f8, fd=17, events=1, cbdata=0x0) at chan_iax2.c:3956 #3 0x0805113c in ast_io_wait (ioc=0x80e7ea8, howlong=1000) at io.c:268 #4 0x4056f3a4 in network_thread (ignore=0x0) at chan_iax2.c:4763 #5 0x400296de in pthread_start_thread () from /lib/libpthread.so.0 (gdb) bt #0 0x62696c2f in ?? () #1 0x0805b2b9 in ast_set_read_format (chan=0x10, fmts=135063712) at channel.c:1482 #2 0x40575b70 in socket_read (id=0x80e98f8, fd=17, events=1, cbdata=0x0) at chan_iax2.c:3956 #3 0x0805113c in ast_io_wait (ioc=0x80e7ea8, howlong=1000) at io.c:268 #4 0x4056f3a4 in network_thread (ignore=0x0) at chan_iax2.c:4763 #5 0x400296de in pthread_start_thread () from /lib/libpthread.so.0 (gdb) bt #0 0x0805befe in ast_translator_free_path (p=0x80ce728) at translate.c:84 #1 0x0805b2b9 in ast_set_read_format (chan=0x810aa58, fmts=135070088) at channel.c:1482 #2 0x40575b70 in socket_read (id=0x80e8a88, fd=17, events=1, cbdata=0x0) at chan_iax2.c:3956 #3 0x0805113c in ast_io_wait (ioc=0x80e7c68, howlong=1000) at io.c:268 #4 0x4056f3a4 in network_thread (ignore=0x0) at chan_iax2.c:4763 #5 0x400296de in pthread_start_thread () from /lib/libpthread.so.0 Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Core dumps from Asterisk
On Tue, 30 Sep 2003, duncan wrote: cvs update Got 3 core dumps from asterisk in a very short period of time, not sure what was going on, but I am posting these in case anybody wants to see them, if you want to see more info I can provide it on request. I am using the asterisk 0.5.0 download. he mentioned he was using the asterisk 0.5.0 download though. surely this means we should update the 0.5.0 release to solve these problems? can you confirm that it was the asterisk-0.5.0.tar.gz file install that caused these segfaults? It was, I just downloaded the tar file, I haven't touched CVS. I assume the tar file is intended to be the latest stable version, unlike a CVS checkout which would give me a development version? Thanks, Michael duncan ___ 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] Flash and Call Transfer
I am new to asterisk and have been playing with it for just a few days. The GrandStream BudgeTone-102 equipment I have allows me to do a blind transfer, but I am interested to know how to do something which isn't blind. I have managed to enable the call parking which allows me to send a call to extension 700 for somebody to collect, but it still seems rather awkward. I also have problems with the 'flash' button, nothing seems to happen when I press it. Strangely though the flash button did start working at one point and I was able to switch between two different calls using it, but rebooting the phones seemed to result in it not working anymore? I would be delighted to read any documentation, but I am struggling to find anything that refers to these issues. I have seen handbook version 2, is there anything else I am missing? Many thanks, Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
