[asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
Do I get 1.4.1 by just doing an svn up in a checkout of the 1.4 branch, or do I have to switch to a new tag or branch for what I have checked out? I did an svn up and there are new files, but nothing in the change files about it being 1.4.1.Many packages with various minor versions tend to have the master branch (like 1.4) mean The latest stable version of 1.4, be it 1.4.0 or 1.4.whatever, while if you check out 1.4.1 that means you stay at 1.4.1 even if there is a 1.4.8. What's the procedure here? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial question
you should separate to two lines, like...
exten = _366[5-9]X,...
exten = 36700,...
Hall, Eric M. wrote:
D
Not sure why this works
exten = _3665[0-9],1,goto(test|${EXTEN}|1)
but this does not.
exten = _366[50-59],1,goto(test|${EXTEN}|1)
I would like to route 36650 – 36700 to a Context ‘test’ however I’m
only able to get 10 to work at a time. Any ideas?
Any help would be great!
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[asterisk-users] Voice quality issues
Being new to Asterisk, I have set up a little test rig at home. Asterisk itself is the Debian Etch package, running on a Celeron 1.7G machine I have two clients on a 100Mb LAN, a Windows XP machine with an Athlon XP2200+ processor, and a Linux Core Duo 6300 machine. On the windows XP machine I run idefisk. On the linux machine I can run either idefisk or kiax. Both of these are obviously IAX clients. When I connect the two of them together, I am getting quite severe sound quality problems in the direction of the linux to windows machine. The best I can describe it is the sound coming out the windows machine sounds like a Darlek (very tinny sound with high pitched echo sounds). I think it represents an over compression problem. The sound coming the other way (windows to linux) is fine. I was worried about my raw capture capability on linux, so I just tried to record and playback directly on the local machine - but the quality here is good. It also appears that the main media stream is not going through the Asterisk server as I have an iptables firewall that is counting iax packets hitting the machine and it is small (in the few tens). How can I debug whats wrong here so I can try and correct it. -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice quality issues
On Sun, Mar 04, 2007 at 11:03:21AM +, Alan Chandler wrote: Being new to Asterisk, I have set up a little test rig at home. Asterisk itself is the Debian Etch package, running on a Celeron 1.7G machine I have two clients on a 100Mb LAN, a Windows XP machine with an Athlon XP2200+ processor, and a Linux Core Duo 6300 machine. On the windows XP machine I run idefisk. On the linux machine I can run either idefisk or kiax. Both of these are obviously IAX clients. Can they be configured to talk directly to one another? When I connect the two of them together, I am getting quite severe sound quality problems in the direction of the linux to windows machine. The best I can describe it is the sound coming out the windows machine sounds like a Darlek (very tinny sound with high pitched echo sounds). I think it represents an over compression problem. Can you try an echo test vs. the Asterisk server? The sound coming the other way (windows to linux) is fine. I was worried about my raw capture capability on linux, so I just tried to record and playback directly on the local machine - but the quality here is good. It also appears that the main media stream is not going through the Asterisk server as I have an iptables firewall that is counting iax packets hitting the machine and it is small (in the few tens). How can I debug whats wrong here so I can try and correct it. What codecs do you use? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
On 01:01, Sun 04 Mar 07, Brad Templeton wrote: Do I get 1.4.1 by just doing an svn up in a checkout of the 1.4 branch, or do I have to switch to a new tag or branch for what I have checked out? I did an svn up and there are new files, but nothing in the change files about it being 1.4.1.Many packages with various minor versions tend to have the master branch (like 1.4) mean The latest stable version of 1.4, be it 1.4.0 or 1.4.whatever, while if you check out 1.4.1 that means you stay at 1.4.1 even if there is a 1.4.8. branches/1.4 is the ongoing 1.4 development tree tags/1.4.1 is the fixed 1.4.1 release So the schema you described above is valid for asterisk as well -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] running error: load_modules: No 'modules.conf' found - vesrion 1.4.1 from svn
Hi, I have just installed the fresh svn version of asterisk and when I run it I get the following errors: [Mar 4 14:19:27] WARNING[24527]: loader.c:728 load_modules: No 'modules.conf' found, no modules will be loaded. [Mar 4 14:19:27] NOTICE[24527]: manager.c:2681 init_manager: Unable to open management configuration manager.conf. Call management disabled. [Mar 4 14:19:27] NOTICE[24527]: cdr.c:1093 do_reload: CDR simple logging enabled. I get this errors although my astetcdir contains the above considered files?? Does asterisk searched for those files in other directory than astetcdir?? the contents of my asterisk.conf: [directories] astetcdir = /home/asterisk/asterisk/asterisk astmoddir = /home/asterisk/asterisk/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astdatadir = /var/lib/asterisk astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk Bests Tomasz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
Brad Templeton wrote: I did an svn up and there are new files, but nothing in the change files about it being 1.4.1.Many packages with various minor versions tend to have the master branch (like 1.4) mean The latest stable version of 1.4, be it 1.4.0 or 1.4.whatever, while if you check out 1.4.1 that means you stay at 1.4.1 even if there is a 1.4.8. You will never see ChangeLog files in the branches in our Subversion repository, because we only create them in the tags as we make releases. Since you didn't give us the output of 'svn info', we don't know what you have already checked out... but if you checked out http://svn.digium.com/svn/asterisk/branches/1.4, then you have everything that is in Asterisk 1.4.1 plus whatever changes have been committed to the 1.4 branch since the release was made. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PRI progress codes.
I assume in this case, you are making the out of the PRI line. Well, that's exactly how PRI works. What you should do is look at the progress code and determine what the call status are (busy, disconnected number, moved number, etc) and play a proper message for the customer. As for the second case, it is not common for PRI lines hold up the call with dead air for 15 to 30 seconds. In my 20 years of dealing with the carriers, it is usually caused by the original PRI network is routing the call to a different network. Usually a different provider and mostly through an analog interface. In this case, there is not a thing you can do about it. Welcome to the true world of telecommunications. -Original Message- From: [EMAIL PROTECTED] on behalf of John Bittner Sent: Fri 3/2/2007 2:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] PRI progress codes. Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a fast busy. I am working with the carrier to get this fixed but its not going easy. Is there anyway when asterisk sees the progress code to cancel the dial and playback a message mapped to the progress code type. Any help on this would be appreciated. John Bittner Simlab.net 9734333011 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users attachment: winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Sending SMS
Steve Well I have 300 for $5.00 thats .016 cents each IF I use all 300 now if I go over I pay .15 each ( I Think) Never went over I see that the cell providers are looking at SMS as internet data over there system and I do agree that there is more money in data than voice services. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Steve Totaro wrote: Gordon Henderson wrote: On Fri, 2 Mar 2007, Al Bochter wrote: I don't see why the cost to send SMS is around .15 each. What does the gateway know that I don't know about sending the SMS. I just think .15 for each SMS send is high. Or am I just over looking something? You're missing nothing; The telcos have us by the short curlys. For them, it's money for old rope. They probably (in the UK at least) make many times more money through TXT messages than voice. The base rate here is about 12p a message. 12p for 160 bytes, or a single data packet over their network - which would be over £700 per MB. There are now bolt ons or additional packages depending on the network you're with - eg. with my contract I get up to 500 free TXTs a month. I know some people who send dozens a day here. (Especially young people - I think most 10 year olds now have mobile phones!). It's scandalous, but no-one challenged it when they first anounced it because we all thought it was fantastic! The best thing they ever did was for the 4 networks (in the UK) to agree to pass TXT messages between each other. That was some 6 or 7 years ago, maybe more, and that's when it really took off big time in the UK. I doubt it'll ever change because that's the way it's always been, and no-one is going to challenge them in a serious fashion. (And no-one else can afford to build up a network to make it possible!) I've not really looked into the TXT sending business via landline in the UK, but I think it's basically a call to an 09xxx number - which are premium rate numbers, charging up to £1.50 a minute. Lets hope the 160 byte packet gets sent in less than a minute! The stats. are amazing too. I looked at wholesale connection last year for a project. They had rates of up to a million messages a month. (do the sums and workout how many miuntes there are in a month...) A quick search shows that in 2004, we in the UK were seding over 20 billion TXT messages a year - Thats 75 million a day. Not bad for a population of 65 million... Who knows what the rate is today... http://www.theregister.co.uk/2004/01/22/uk_text_message_volumes_break/ Ah, that was 2004. Looks like we're almost doing that per month: http://www.theregister.co.uk/2006/06/26/uk_sms_record/ 3.3 billion texts sent in May 2006... Gordon Text messaging is not that big in the US for some reason. Well anyways, on my T-Mobile phone, I have an unlimited text message package that cost $15/mo. I am not sure how many constitutes unlimited though, I have not read the small print. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000721-1, 03/03/2007 - 3/3/2007 9:05:20 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] When does local leg in call file start?
For a simple call file like Channel: Zap/g1/XXX RetryTime: 60 WaitTime: 30 Context: from-file Extension: s Priority: 1 I noticed that [EMAIL PROTECTED] started to execute regardless of the state of the outgoing call. Is this supposed to be? So far I can only set a Wait() in the local leg and hope the remote party picks up soon enough. I thought call file extension will start execution only when the outgoing leg is answered. Or is there some way to detect this? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does local leg in call file start?
Yuan LIU wrote: For a simple call file like Channel: Zap/g1/XXX RetryTime: 60 WaitTime: 30 Context: from-file Extension: s Priority: 1 I noticed that [EMAIL PROTECTED] started to execute regardless of the state of the outgoing call. Is this supposed to be? So far I can only set a Wait() in the local leg and hope the remote party picks up soon enough. I thought call file extension will start execution only when the outgoing leg is answered. Or is there some way to detect this? FXO ports are considered answered as soon as dialing is finished. This is because most telcos do not provide answer supervision. PRI (and maybe EM ports) and FXS ports do not have this issue. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does local leg in call file start?
Yuan LIU wrote: I noticed that [EMAIL PROTECTED] started to execute regardless of the state of the outgoing call. Is this supposed to be? So far I can only set a Wait() in the local leg and hope the remote party picks up soon enough. I thought call file extension will start execution only when the outgoing leg is answered. Or is there some way to detect this? If you are dialing via a PRI or a device that supports call supervision, this is the case. If you are using a standard POTS line, the call is assumed answered immediately. This has been covered many times on this list, search the archives for code fragments on how to deal with such a situation. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] x100p.com
Is this site good ? They ship on time ? Any reviews on their card . Thx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
On Sun, Mar 04, 2007 at 08:50:48AM -0600, Kevin P. Fleming wrote: Brad Templeton wrote: I did an svn up and there are new files, but nothing in the change files about it being 1.4.1.Many packages with various minor versions tend to have the master branch (like 1.4) mean The latest stable version of 1.4, be it 1.4.0 or 1.4.whatever, while if you check out 1.4.1 that means you stay at 1.4.1 even if there is a 1.4.8. You will never see ChangeLog files in the branches in our Subversion repository, because we only create them in the tags as we make releases. Since you didn't give us the output of 'svn info', we don't know what you have already checked out... but if you checked out http://svn.digium.com/svn/asterisk/branches/1.4, then you have everything that is in Asterisk 1.4.1 plus whatever changes have been committed to the 1.4 branch since the release was made. Thanks, Kevin. Yes, I have http://svn.digium.com/svn/asterisk/branches/1.4. I was running /trunk before but it wasn't stable enough to be a production system (no surprise.) I am presuming that the above is intended to be stable in this fashion. In many packages there is some file (usually the change log) which always tells you what version of the program you have in your hands, in terms of the program's current version number -- of course you can see the svn revision numbers and dates but they don't trivially translate. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
Brad Templeton wrote: In many packages there is some file (usually the change log) which always tells you what version of the program you have in your hands, in terms of the program's current version number -- of course you can see the svn revision numbers and dates but they don't trivially translate. I would be surprised to see such a file in a direct checkout from the project's SCM system. Even if that file existed, it would exist for a very short time as the moment a new commit occurred that branch would no longer 'be' 1.4.1, for example. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Real Time, sip.conf, [general]
Hi, I am implementing de Real Time architecture, I would like to know if, their is any problem in putting the section [general] of the sip.conf file in the table of sippeers. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] running error, unable to load *.conf files: load_modules: No 'modules.conf' found - svn version 1.4.1
Hi All! I have just installed the fresh svn version of asterisk and when I run it I get the following errors: [Mar 4 14:19:27] WARNING[24527]: loader.c:728 load_modules: No 'modules.conf' found, no modules will be loaded. [Mar 4 14:19:27] NOTICE[24527]: manager.c:2681 init_manager: Unable to open management configuration manager.conf. Call management disabled. [Mar 4 14:19:27] NOTICE[24527]: cdr.c:1093 do_reload: CDR simple logging enabled. I get this errors although my astetcdir contains the above considered files?? Does asterisk searched for those files in other directory than astetcdir?? When I for instance try to reload the chan_sip.so through the CLI interface I get same that sql.conf file is unable to be opened. the contents of my asterisk.conf: [directories] astetcdir = /home/asterisk/asterisk/asterisk astmoddir = /home/asterisk/asterisk/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astdatadir = /var/lib/asterisk astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk Thanks in advance! Bests Tomasz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
On Sun, Mar 04, 2007 at 02:34:21PM -0600, Kevin P. Fleming wrote: Brad Templeton wrote: In many packages there is some file (usually the change log) which always tells you what version of the program you have in your hands, in terms of the program's current version number -- of course you can see the svn revision numbers and dates but they don't trivially translate. I would be surprised to see such a file in a direct checkout from the project's SCM system. Even if that file existed, it would exist for a very short time as the moment a new commit occurred that branch would no longer 'be' 1.4.1, for example. Yup, typically it's a changelog and significant changes are noted along with version number bumps. I'm presuming the /branch/1.4 is the latest stable version of 1.4 with the latest patches.Since there is a 1.4.1 it means it is also 1.4.1 with the latest patches -- or so I presume. Having a file means people can look at see what they have, without having to ask here :-) Now that I know I can interpret what it means.I'm assuming that the latest /branch/1.4 is the one to run if you want a stable system with all known and tested patches and fixes but only modest new functionality -- or should one really be running /tags/1.4.1 and regularly updating your tag in order to get that? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
Brad Templeton wrote: Yup, typically it's a changelog and significant changes are noted along with version number bumps. I'm presuming the /branch/1.4 is the latest stable version of 1.4 with the latest patches.Since there is a 1.4.1 it means it is also 1.4.1 with the latest patches -- or so I presume. The 1.4 branch is where we continue putting bug fixes as issues arise. As we do releases we make a snapshot of the branch, which becomes a tag. So 1.4.1 is a snapshot of the 1.4 branch at a specific revision. Having a file means people can look at see what they have, without having to ask here :-) Now that I know I can interpret what it means.I'm assuming that the latest /branch/1.4 is the one to run if you want a stable system with all known and tested patches and fixes but only modest new functionality -- or should one really be running /tags/1.4.1 and regularly updating your tag in order to get that? If you are using a tag checkout an update won't get you anything as it's a snapshot that isn't modified. If you are running the 1.4 branch then an update will get you the latest bug fixes. Note that the 1.4 branch *only* gets bug fixes. It does not get new functionality. Hopefully this enlightens you a bit. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] running error, unable to load *.conf files: load_modules: No 'modules.conf' found - svn version 1.4.1
TZieleniewski wrote: I have just installed the fresh svn version of asterisk and when I run it I get the following errors: What is the 'fresh svn' version? There are at least 30 different flavors of Asterisk that can be checked out from Subversion. A 'show version' might give us a clue what you are working with. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX using T38
I'll consider the offer if it includes your code being included with Asterisk. On 3/3/07, Steve Underwood [EMAIL PROTECTED] wrote: I'll do it for 30% less than they quote. :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium S101i - pickupexten doesn't work
It is my understanding that pickup does not work across channel technologies and Sipura does not support IAX. You can use the pickup() application however. On 3/1/07, Joseph [EMAIL PROTECTED] wrote: How to configure Digium S101i adapter to work with pickupexten *8 ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium S101i - pickupexten doesn't work
My understanding is that chan_iax2.c does not support call pickup. I am pretty sure pickup works between things like SIP and ZAP. Andrew Joakimsen wrote: It is my understanding that pickup does not work across channel technologies and Sipura does not support IAX. You can use the pickup() application however. On 3/1/07, Joseph [EMAIL PROTECTED] wrote: How to configure Digium S101i adapter to work with pickupexten *8 ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] running error, unable to load *.conf files: load_modules: No 'modules.conf' found - svn version 1.4.1
Aterisk SVN-branch-1.4-r57649M one more thing I run ./configure with the following parameters: --prefix=/home/asterisk --exec-prefix=/home/asterisk/asterisk --sysconfdir=/home/asterisk/asterisk Cheers tomasz Kevin P. Fleming napisa(a): TZieleniewski wrote: I have just installed the fresh svn version of asterisk and when I run it I get the following errors: What is the 'fresh svn' version? There are at least 30 different flavors of Asterisk that can be checked out from Subversion. A 'show version' might give us a clue what you are working with. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configurations Files of TE110P
please can someone send to me his files like zaptel zapta if he si using TE110P thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Java w/ Threads
Let me rephrase that: It would be a lot *simpler*. From the programming point of view you are connecting to a single events source. Astmanproxy is very stable as well. On 3/3/07, Stefan Reuter [EMAIL PROTECTED] wrote: Jesus Mogollon wrote: The best option would be to use AstManProxy and connect your event manager to it. why would adding a new system in between be better than directly connecting to multiple Asterisk servers? =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] Steuernummern 215/5140/1791 USt-IdNr. DE220701760 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX using T38
Andrew Joakimsen wrote: I'll consider the offer if it includes your code being included with Asterisk. On 3/3/07, Steve Underwood [EMAIL PROTECTED] wrote: I'll do it for 30% less than they quote. :-) So, you want a discounted price for something that offers more than Attractel will offer you? Do you understand how negotiation works? :-) Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX using T38
Steve Underwood wrote: So, you want a discounted price for something that offers more than Attractel will offer you? Do you understand how negotiation works? :-) He certainly understands from the customer's point of view G ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configurations Files of TE110P
Here you go. This is from ATP http://www.austechpartnerships.com/forum/viewtopic.php?t=76 /etc/zaptel.conf loadzone=au defaultzone=au span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 dchan=16 This assumes connection to a carrier, where they provide clocking. /etc/asterisk/zapata.conf switchtype = euroisdn signalling = pri_cpe group = 1 pridialplan=unknown context = incoming channel = 1-15 channel = 17-31 Cheers, Joel On Sun, 2007-03-04 at 23:41 +, younss azzayani wrote: please can someone send to me his files like zaptel zapta if he si using TE110P thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail to SMS using asterisk
1. I want my users to dial certain number. 2. Record a voicemail with destination number. No problems so far 3. Convert this Voicemail to Text. Save the VM in database, but you need to build the speech-to-text system somehow. The database will make the external application easier to access the data. 4. Send the text with sms apps. and I wish i connect my asterisk to smsc directly. Is it possible without kannel? Then how you plan to send the SMS? There are plenty of ways but you need *SOME* interface be it kannel, FastSMS, GSM modem, email, etc, etc, etc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple simultaneous calls
There is also an Ethernet/SIP overhead speaker. Voipsupply sells it. On 3/2/07, Stefano Totaro [EMAIL PROTECTED] wrote: Quite surprising, yes! :-) I am from north east Italy, now I live in Verona (Romeo and Juliet's city :). I cannot do it connecting amp to the PBX. I have quite a long distance to cover and a network is already there. My phones are have a quite smart processor so we may probably run the ices2 that you are suggesting or something similar. I will check the links that you sent me. Thanks, Stefano Stefano Totaro, Off topic. I just noticed your name and was a little surprised!? ;-) Are you in Italy / Sicily? Anyways, you can achieve overhead paging through a sound card hooked to an Amp and speakers from your PBX. I have yet to do it but have read about it. I think this may be the better solution for you unless you are set on doing it over IP. Check here for several options http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom I am pretty sure if you use a ring group or meetme, there is no way around each phone having it's own stream. Interestingly, 3Com systems do conferencing and paging through multicast which is a nice idea but in practice can be a real pain to configure network components to work properly (especially if you do not control the network or you are trying to implement paging between remote offices). I have spent hours on this exact problem in the past. If it were me, I would probably not want all that traffic on the PBX unless that is all that it will be doing or if you go the sound card route. I would use ices2 and let everyone stream from a different server than the PBX. Since you are using phones, I do not know that ices2 would work for you, something must initiate the call. I would probably have a second Asterisk box to just handle the paging, setup an extension the dialplan of the main PBX to dial the paging machine via SIP (and possibly include Authenticate) that would drop the call into something like this: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page Thanks, Steve Totaro stefano.totaro at transport.alstom.com wrote: Hello Steve, thanks for your anwer. Yes, you are right we want to do VoIP telephone system capable also of public address (overhead paging) service. So synchronization is a key issue if we want to avoid unpleasant effects. We are designing our phones and they will have also onboard amplifiers. What I am trying to understand is whether we may use the phone system also for this service or if it is better to go for a specific streaming technology (Ices2 is a good suggestion thanks). What happen if I put all the phones in a ring? Do they join the same multicast stream or a single stream for each phone will be created? Thanks again. Stefano Inactive hide details for Steve Totaro stevetotaro at hotmail.comSteve Totaro stevetotaro at hotmail.com *Steve Totaro stevetotaro at hotmail.com* Inviato da: asterisk-users-bounces at lists.digium.com Phone: 01/03/2007 18.33 Per favore, rispondere a Asterisk Users Mailing List - Non-Commercial Discussion Per: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users at lists.digium.com Cc: (ccr: Stefano TOTARO/ITVRN01/Transport/ALSTOM) Oggetto: Re: [asterisk-users] Multiple simultaneous calls stefano.totaro at transport.alstom.com wrote: Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I have to open 50 connections and send 50 times the same packets or if can use in some way the multicast. Is there anybody that may give me some idea. Thanks in advance, Stefano I suppose you could do that although, I am unclear on the auto-answering softphone and the loudspeaker thing. Is this just for overhead paging or something? You could put all the phones in a ring group with ringall and use the computer's sound card to connect to an amplified speaker setup. You could also look at ices2 to stream audio or some other streaming technology. Thanks, Steve Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com --
Re: [asterisk-users] Configurations Files of TE110P
thank's Joel, i've a problem with the card TE110P the Teleco provider told me that to work with a corssover cable 1-4 2-5, but the led still red i don't know why? , i a fear that if i play with the cable i dammage my card or the teleco modem card so i'm lost :) thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configurations Files of TE110P
http://www.austechpartnerships.com/forum/viewtopic.php?t=76 from the link you give me i see RJ45 pins used for E1 is 1,2,4,5 - which you may need to cross (note you can't use standard CAT5 cables here for that). what's that mean, is it mean that i have to use cable CAT6 or what exactly Can someone Give me A good schema of how to cross this cable for me 14 25 41 52 but this doesn't work ,,, :-( :-( :-(( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [A*UG] How to log VERBOSE statement to a file?
Larry Alkoff wrote: I would like to log a verbose statement in my 900/976 extens to a special file called 'attacks'. These are not standard messages like debug, notice, warning, error, vebose or dtmf that could be logged to /var/log/asterisk/messages. Does the 'verbose' in VERBOSE commands have anything to do with the 'verbose' in error messages? I tried redirection of a VERBOSE statement - did not work. Larry Replying to myself. By experiment I found that you could include verbose on the on the messages line in logger.conf. In my case, I simply added a line verbose=/var/log/asterisk/verbose In the case of serious breeches I can grep attack: verbose Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does local leg in call file start?
From: Doug Lytle [EMAIL PROTECTED] Date: Sun, 04 Mar 2007 13:56:35 -0500 Yuan LIU wrote: I noticed that [EMAIL PROTECTED] started to execute regardless of the state of the outgoing call. Is this supposed to be? So far I can only set a Wait() in the local leg and hope the remote party picks up soon enough. I thought call file extension will start execution only when the outgoing leg is answered. Or is there some way to detect this? If you are dialing via a PRI or a device that supports call supervision, this is the case. If you are using a standard POTS line, the call is assumed answered immediately. This has been covered many times on this list, search the archives for code fragments on how to deal with such a situation. Doug Thanks for the explanation, Doug and Eric. Yes I came across those threads several times, just didn't quite relate to call files. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is the 1.0.X branch vulnerable to the SIP issue?
We are still using 1.0.7 and did not see any patches for the 1.0.X branch. Does anyone know if that branch is affected? -Thermal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Read() status?
Does application Read() return a status? Console displays stuff, but show application read doesn't mention any status variable. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
