Re: [Asterisk-Users] PHP Manager examples
CW_ASN wrote: Here is my example. I'm using a lot of times a day. ?php $socket = fsockopen(192.168.0.53,5038, $errno, $errstr, $timeout); fputs($socket, Action: Login\r\n); fputs($socket, UserName: admin\r\n); fputs($socket, Secret: blabla\r\n\r\n); fputs($socket, Action: Command\r\n); fputs($socket, Command: reload\r\n\r\n); $wrets=fgets($socket,128); ? Thank you! Added to http://www.voip-info.org/tiki-index.php?page=Asterisk+manager+API /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recommended places for beginner to start?
Actually asterisk.org has all the info you need. Just install the linux distrib with CVS, kernel sources and openssl-devel and all their dependencies. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew England Sent: Monday, November 03, 2003 3:56 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Recommended places for beginner to start? (The list may get this msg twice; I originally sent it from the wrong email address, my apologies. Moderator, if you can, please delete my original email submission from [EMAIL PROTECTED] Thanks.) Hello- Summary: Can anyone recommend a place to start to learn how to create an Asterisk system given a basic Digium PCI card and some generic PC hardware? Details: I plan to help a friend not familiar with Linux platforms load and evaluate a Digium/Asterisk system for business-development purposes. A couple years ago I used to work as a Unix/Linux sw developer and sysadmin, but have been doing sales/marketing stuff since. Where should I start to read about loading a system? My friend apparently has a $100-flavor of Digium for eval purposes (can hook up to one external phone line, or so I'm told), but knows little else. Since I've been the unix/linux geek in a past life, he came to me for assistance. I downloaded the .pdf handbook, and their appeared to be a reference to a downloading and installing section, but I couldn't find any text/body that actual described this process. Do I pick any linux flavor (presumably with compatible kernel) like RedHat/Debian/SuSE and load up the source/pkgs/rpms necessary and let 'er rip? Will I get a phone switch/PBX (or whatever this is) going fairly easily, assuming I get my linux box/platform fired up ok? Any gotchyas, tricks of the trade, things to know/worry about, etc? Is this all contained in the .pdf handbook? When I skimmed it, I didn't find anything that seemed to match up with a installing for a rookie's perspective like mine, but maybe I overlooked something. I have yet to get my hands on the Digium hardware/docs/etc that my buddy ordered; maybe some answers/secrets/support-resources are in there? I'm on vacation right now and am a little short on info, but before delving into this when I get back (probably starting around 11/5) I thought I would send out this note to the user list so that I might potentially save some time in research/pain before I start. Thanks for any help! -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie Questions
Look into www.digium.com. Digium's cards are you best choice. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brez Sent: Monday, November 03, 2003 4:03 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie Questions hello, I am completely new to things but was wondering if some one could steer me in the right direction [i.e. i was volunteered to get a PBX running with little or knowledge] good news is, i got a lot of experience with open source / linux / etc. anyhow. we have 4 lines coming in and need 16 extensions. we have the PC and the 16 analog phones. the question is what type of hardware will i need? i.e. modem, a phone 'hub' [or whatever it is called for pluggin all the phone lines into] - basically a small office environment. if any of you using asterisk in a similar environment could spell out exactly what hardware youre using [and perhaps where to buy it] for your office, i would really appreciate the help. thanks. ___ 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] Asterisk behind LinkSys NAT Routing
Robert Mann wrote: Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside and outside through * is not possible at all. I know I can not be the only one who has tried to do this. Please any help would be greatly appreciated. Robert, You need to get Asterisk onto a public IP address.. Using the DMZ function on the router will not work.. If you search the archives you will see that it has been attempted many times.. The reason is not in the IP but in the SIP headers.. they will be sent out from the Asterisk server with the internal IP address of the server, this means that when the SIP UA reads the SIP message and responds it will respond to the incorrect IP address.. So the basic rules where NAT is involved are.. Asterisk server must always be on a public IP address.. SIP UA's can be behind NAT but need nat=yes, canreinvite=no and qualify=yes set in the phone configuration in sip.conf.. Hope that helps.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi Brian, - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, November 02, 2003 11:54 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) ... Its a start but having to restart when you change registration isn't very intuitive. But its an excellent point to start. Good luck. I know it... it will be solved in the next release... no need to restart application. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
WipeOut wrote: So the basic rules where NAT is involved are.. Asterisk server must always be on a public IP address.. You keep saying this, but it is not correct. I have several asterisk servers running behind NAT servers, and they function perfectly. I won't say configuring them was as easy as doing the ones on public IPs, but it is not impossible, just tricky. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Please provide your feedback about the application Only in that way it can be improoved. Thanks! Dan - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 12:13 AM Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform) Finaly, someone has started the IAX soft phone ball :) Thanks, Dan... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another newbie question
Thanks Jose/Tom for responding to my Newbie questions. its much clearer now. anyhow on to the next [unrelated question] here's the use case: i will need one machine that will answer incoming calls - store the caller's number [caller ID] and then prompt the caller to answer a question by using the dialpad [e.g. please enter your zip code] and then store all the information in MySQL [or any persistant storage will do] what i got so far: it looks like digium Wildcard X100p will answer the phone and get the caller's number [caller ID] but my remaining question is: can i have the caller respond to question [by pressing the dial pad] and can i store that information [and the caller ID] somewhere? any suggestions would be greatly appreciated. thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi , - Original Message - From: Masakazu Nakano [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 5:05 AM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) ... thanks for good application! and I wish 'no with installer' package about that. because I think use with USB-memory device in any places (ie.net-cafe.) The next release (which will be available during this week) will be in two versions: one with installer and one wich will be possible to run it from any type of media. It has just 4 files (main exe, a dll, one configuration file and one call list file). is that need registry setting or not? No.. In the actual version (with installer), just the installer writes something in the registry in order to be able to uninstall. The DLL does not need to be registered (it si not an Active X dll) Nothing used by the application stored in the system registry. This was one of my first goals... to keep it as portable as possible Even with the actual version, you can make a small BAT file to copy the 'wiax.dll' file in the system32 directory before starting the application...this is all. Nothing to register. All the required files for the application to run are: diax.exe, diax.cfg, diax.cl (all in the same directory) wiax.dll (in the system32 directory). I even think to avoid using an installer mainly because the installer part is bigger that the application himself. What do you think? Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a bit frightened, guys
On Sun, Nov 02, 2003 at 06:20:15PM -0500, Andrew Kohlsmith wrote: So here's my humble request. Can someone who has implemented a live production Asterisk deployment, preferably between two sites (HQ and a branch office, connected over the internet) spare the time and contact me here, or to my email directly? As a lurker, I would very much appreciate if this conversation could be kept on-list. Not only does it help more than just yourself then, but it also gets to be part of the archive which search engines can access. Certainly, I too am trying to push for an Asterisk-based solution at my workplace rather than a proprietary one, and some real-world cases would be a great boost rather than just saying to management 'yes it's worked well in loads of places' - the next question is for who? where? show me details and reports. At present I am totally unable to answer these questions because of the lack of available information :( So, if any such discussion could be kept on the list, then the knowledge and experience can be shared amongst all, in the same spirit that the Asterisk software has been :) Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Another newbie question
Look into AGI, there a re some examples out there, but it's very much doable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brez Sent: Monday, November 03, 2003 11:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Another newbie question Thanks Jose/Tom for responding to my Newbie questions. its much clearer now. anyhow on to the next [unrelated question] here's the use case: i will need one machine that will answer incoming calls - store the caller's number [caller ID] and then prompt the caller to answer a question by using the dialpad [e.g. please enter your zip code] and then store all the information in MySQL [or any persistant storage will do] what i got so far: it looks like digium Wildcard X100p will answer the phone and get the caller's number [caller ID] but my remaining question is: can i have the caller respond to question [by pressing the dial pad] and can i store that information [and the caller ID] somewhere? any suggestions would be greatly appreciated. thanks ___ 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] NetJet Cards
Hi Matthew,
On 2 Nov 2003, Matthew Enger wrote:
exten = _004,1,Dial(modem/g1/V${EXTEN:1})
Try this Dial command:
Dial(Modem/ttyI0:${EXTEN:1})
msn=0397468733L*
Try removing L* from the MSN, it looks wrong to me.
You might find that ttyI0 and ttyI1 are both channels of the first card,
and that you will need ttyI2 and ttyI3 for the second card. But I haven't
tested isdn4linux with more than one card.
Please let me know if you get it working, I had major problems with
Asterisk hanging using the isdn4linux and other modem drivers. Eventually
I had to switch to CAPI.
Cheers, Chris.
--
___ __ _
/ __// / ,__(_)_ | Chris Wilson -- UNIX Firewall Lead Developer |
/ (_ / ,\/ _/ /_ \ | NetServers.co.uk http://www.netservers.co.uk |
\ _//_/_/_//_/___/ | 21 Signet Court, Cambridge, UK. 01223 576516 |
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Re: [Asterisk-Users] High Availability and Mass Deployment for Asterisk
On 31/10/03 12:11, Senad Jordanovic wrote: You are right, but what if each * server had a single source for all of its configuration files from a file server over NFS or similar. Single point of failure at the file server. Better to rsynch all the machines config files or similar. -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
I cannot seem to get the software to work on my machine. I am multihomed running windows XP home. Perhaps the software is binding to the card not connected to asterisk. If I turn on debugging in asterisk I see no IAX stuff coming in from the IP. Thanks, Will - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 3:21 AM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Please provide your feedback about the application Only in that way it can be improoved. Thanks! Dan - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 12:13 AM Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform) Finaly, someone has started the IAX soft phone ball :) Thanks, Dan... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS What to do?
If your DSL link is the bottleneck, rather than earlier hops back through the providers network, the provider could also prioritize VOIP packets going up the DSL line. That requires a cooperating provider, of course. You may also setup a linux box (or another QoS supporting router) on the inside and tune the communication with queueing there. Read the LARTC howto for more info. roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
Shoval Tom wrote: Isn't putting asterisk on the public IP network a bad idea? Is it a bad idea?, Not really if you take the right precautions..From how you described your setup you have connected your server directly to the internet anyway.. If you nominated you Asterisk box as the DMZ host in your router it effectively is directly on the internet.. if you havent secured the box itself I suggest you do.. :) What about security? This is somthing that you will need to take care of.. Of course some people's opinions on securing a PC is to not connect it to the internet at all, of course that is a little silly.. You will have to decied on the level of security you are happy with.. This is a topic that can be debated for days so I will not get into it any further than that.. And how will all us newbies make the linux box as secure as possible? The quickest way is to setup an IPTABLES firewall.. You will need ports 5060 and 1 to 2 open for a default Asterisk install using SIP only.. (NOTE: make sure you know how to activate and deactivate IPTABLES from a command line because while you are playing there is a good chance you will lock yourself out of the server from any remote PC and you can even break Xwindows running locally with a firewall..) Later.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Monday, November 03, 2003 11:05 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing Robert Mann wrote: Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside and outside through * is not possible at all. I know I can not be the only one who has tried to do this. Please any help would be greatly appreciated. Robert, You need to get Asterisk onto a public IP address.. Using the DMZ function on the router will not work.. If you search the archives you will see that it has been attempted many times.. The reason is not in the IP but in the SIP headers.. they will be sent out from the Asterisk server with the internal IP address of the server, this means that when the SIP UA reads the SIP message and responds it will respond to the incorrect IP address.. So the basic rules where NAT is involved are.. Asterisk server must always be on a public IP address.. SIP UA's can be behind NAT but need nat=yes, canreinvite=no and qualify=yes set in the phone configuration in sip.conf.. Hope that helps.. 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recording files for menues
Hello Olle, Olle, I can't reach the faq page, and haven't been able to for the last four days. I'm getting 504 gateway timeout errors. Gateway timeout indicates something with your web proxy ...or? I've been able to reach the Wiki all weekend, I've updated and created several pages... I also now that Jim have been working to speed things up, among them adding more SQL connections as we have had many hits at the same time when mailing a URL to the list... You should be able to reach it. Is it only this web site or do you get that error message somewhere else? I have not been able to reach www.voip-info.org as well using my default settings. Upon researching the problem I tried nslookup www.voip-info.org which returns an IP address of 192.168.168.3 which is obviously wrong. This is the answer from my local DNS server contacting the t-online DNS server. Upon doing a nslookup to a different DNS server, ie. nslookup www.voip-info.org 141.1.1.1 I get back 64.65.102.50. It seems that your DNS configuration might be broken. rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
IAX2 Java library (was Re: [Asterisk-Users] New IAX software phone (for WIndows platform))
On 03/11/03 00:25, Mark Spencer wrote: As a side note, I strongly would like to see someone implement a client using libiax2 which implements IAX2 instead of the (now obsolescent) IAX version 1. I'm implementing a Java-based IVR server (and yes, I know Asterisk does IVR, and no, it's not flexible enough to do what I want and no, it doesn't integrate well with the Java systems we have, etc. hence my doing this). Currently it uses SIP (using the NIST JAIN-SIP stack) and JMF to handle RTP/audio stuff. I've found that JMF/RTP doesn't scale very well, as it spawns a *lot* of threads, and can't reliably handle more than 20 simultaneous calls. So, I'm investigating the possibility of writing an IAX library for Java. Searching the archives, it seems various other people would be interested in this. So, my questions are: - Should I implement IAX or IAX2? What's the main difference, other than IAX2 supporting trunking (which according to the docs needs a Zaptel timing source). - Has anyone else made any headway with this? - Is anyone else interested in making this an LGPL or even a GPL project and helping me with it? I'm likely to implement just the call management/DTMF/audio type stuff required for IVR initially (i.e. not worry about call xfer, etc.). It'll also be geared towards handling the hundreds of simultaneous calls required in a server environment, although there'll be no reason not to use it for IAX clients too. Obviously such a library would enable a nice GUI cross-platform IAX(2?) client to be easily created, which would be a nice by-product. -- Alastair Maw MX Telecom http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi Will, - Original Message - From: William Carlson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 12:31 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) I cannot seem to get the software to work on my machine. I am multihomed running windows XP home. Perhaps the software is binding to the card not connected to asterisk. If I turn on debugging in asterisk I see no IAX stuff coming in from the IP. Thanks, Will In order to see something in the * console you must register first. Have you enter your credentials and * server IP address when asked? If not registered, nothing works and the application closes by himself. Please give me more details about this behaviour. It must work on a multihomed computer too if a correct route exists to the Asterisk server. If you can ping it, then it is only a registering problem. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NetJet Cards
Hello,
With help from Adam, I managed to get it working. I have put in the bits
which might help others in the future below:
Kernel:
I had to compile a fresh kernel source and apply the voice patch
available from www.traverse.com.au.
Since I had 2 cards, I did a modprobe in my boot scripts of the
following:
/sbin/modprobe hisax type=20,20 protocol=2,2 id=HiSax
modem.conf:
[interfaces]
context=remote
driver=i4l
stripmsd=0
dialtype=tone
mode=immediate
group=1 ; group=1,2,3,9-12
msn=0397468733L*
stripmsd=0
device = /dev/ttyI0
device = /dev/ttyI1
This seems to work for both incoming calls and outgoing calls. The only
issue I still seem to be having but have not had much chance to
investigate is pressing buttons to respond to menus on calls made out.
But I am not sure if it is the config, phone or user who is the problem
yet:)
Extensions.conf:
exten = _004,1,Ringing
exten = _004,2,Dial(modem/g1:${EXTEN:1})
exten = _004,3,Congestion
this assumes that you are using 0 to indicate an outgoing call and that
the number to dial begins with 04 (australian mobile). I found I needed
the Ringing bit in front of dial so that the user knew that the number
was dialing (although it does slow down dial out by 2 or so seconds).
Adam pointed out that DMTF stuff showing up from kernel: (dmesg)
dtmf: tt='1'
which I might want to try and turn off. Have not looked into this
further.
I hope this helps.
Thanks,
Matthew Enger
[EMAIL PROTECTED]
On Mon, 2003-11-03 at 21:24, Chris Wilson wrote:
Hi Matthew,
On 2 Nov 2003, Matthew Enger wrote:
exten = _004,1,Dial(modem/g1/V${EXTEN:1})
Try this Dial command:
Dial(Modem/ttyI0:${EXTEN:1})
msn=0397468733L*
Try removing L* from the MSN, it looks wrong to me.
You might find that ttyI0 and ttyI1 are both channels of the first card,
and that you will need ttyI2 and ttyI3 for the second card. But I haven't
tested isdn4linux with more than one card.
Please let me know if you get it working, I had major problems with
Asterisk hanging using the isdn4linux and other modem drivers. Eventually
I had to switch to CAPI.
Cheers, Chris.
--
Matthew Enger [EMAIL PROTECTED]
Xintegration
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Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
I did set it up to register here is my iax.conf config. [blah] type=friend user=blah secret=blah context=default host=192.168.5.200 This is what I am seeing in asterisk. NOTICE[32773]: File chan_iax.c, Line 2708 (register_verify): Peer 'blah' is not dynamic (from 192.168.5.200) Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: REGREQ Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: REGREJ Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: REGREQ Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: ACK Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: REGREQ Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: REGREJ etc Ok I figured it out I need to change the host field to host=dynamic Thanks, Will - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 5:51 AM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Hi Will, - Original Message - From: William Carlson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 12:31 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) I cannot seem to get the software to work on my machine. I am multihomed running windows XP home. Perhaps the software is binding to the card not connected to asterisk. If I turn on debugging in asterisk I see no IAX stuff coming in from the IP. Thanks, Will In order to see something in the * console you must register first. Have you enter your credentials and * server IP address when asked? If not registered, nothing works and the application closes by himself. Please give me more details about this behaviour. It must work on a multihomed computer too if a correct route exists to the Asterisk server. If you can ping it, then it is only a registering problem. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi, - Original Message - From: William Carlson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 1:15 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) I did set it up to register here is my iax.conf config. [blah] type=friend user=blah secret=blah context=default host=192.168.5.200 This is what I am seeing in asterisk. .. Ok I figured it out I need to change the host field to host=dynamic Thanks, Will This is what I have in the iax.conf file: [yourusername] type=friend username=yourusername secret=blahblah auth=plaintext host=dynamic callerid=Your User Name your user extension context=yourcontext Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
I made few successful calls (in/out). However, the application did crash few times during conversation, and now while trying to start it the application shows this error message: Run Time error '341': Invalid control array index I am using XP-PRO, Service pack 1. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a bit frightened, guys
So here's my humble request. Can someone who has implemented a live production Asterisk deployment, preferably between two sites (HQ and a branch office, connected over the internet) spare the time and contact me here, or to my email directly? As a lurker, I would very much appreciate if this conversation could be kept on-list. Not only does it help more than just yourself then, but it also gets to be part of the archive which search engines can access. Certainly, I too am trying to push for an Asterisk-based solution at my workplace rather than a proprietary one, and some real-world cases would be a great boost rather than just saying to management 'yes it's worked well in loads of places' - A couple of items to consider (in addition to the technical * implementation issues) are: 1. end-to-end connectivity (within each location) as most corp hubs/switches are not on ups, shared devices located under someone's desk where the power cord is kicked, an employee's space heater trips a breaker. 2. legal issues (what happens when an employee needs to call emergency personnel and the phone system doesn't work for whatever reason) 3. how will you deal with QoS issues when they pop up? (someone decides to backup their fixed disk across the local net; the latest virus/trojan is consuming all available bandwidth; user drag/drops very large directory or files.) 4. your ISP decides to block a range of ports and didn't tell you; what's the backup plan and how quickly can it be operational 5. are there requirements for a primary backup * system, and should this be configured with some automated failover process or left to support personnel to handle manually 6. should you have a formal change control process and how does it apply to downloading cvs updates that break production * boxes 7. are there any business requirements for backup support personnel should you get hit by a bus on the way home from work 8. its not uncommon for infrastructure personnel (eg, switch, routers) to reboot, swap out, upgrade, etc, stuff for various reasons. Is there a need to treat those support requirements different when mgmt is accustomed to phone systems being operational 99.999% of the time? 9. should your plan to implement * include a phase-in approach where only a small part of the business is impacted before moving to the next phase? All of those can be easily handled but often times they are not addressed or properly researched up front. Impress your mgmt. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
WipeOut wrote: Shoval Tom wrote: And how will all us newbies make the linux box as secure as possible? The quickest way is to setup an IPTABLES firewall.. You will need ports 5060 and 1 to 2 open for a default Asterisk install using SIP only.. Visit the Wiki page http://www.voip-info.org/tiki-index.php?page=Asterisk+security where you'll find some information about security in Asterisk. (And some missing pages, which I invite other mailing list readers to write!) There's a pointer on that page to a page with a suggested IPTABLES setup. And, as WipeOut stated, make sure you are aware of what you're doing when managing IPTABLES. Of course, no one else than Wipeout have locked himself from the system, no way, we're pro's ;-) hrmm Regards, /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi, - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 1:39 PM Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform) I made few successful calls (in/out). However, the application did crash few times during conversation, ... Have you seen something else related to this crash? It crashes or it just hang up the conversation/ ...and now while trying to start it the application shows this error message: Run Time error '341': Invalid control array index I'll try to see if I can reproduce this behaviour... I am using XP-PRO, Service pack 1. Same for me. Best regards, Dan P.S. I'll post later today a new prerelease (0.9.1) with some bug fixes and some users requested improovements. Keep on eye on this list! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recording files for menues
Peer Oliver schmidt wrote: I have not been able to reach www.voip-info.org as well using my default settings. Upon researching the problem I tried nslookup www.voip-info.org which returns an IP address of 192.168.168.3 which is obviously wrong. This is the answer from my local DNS server contacting the t-online DNS server. Upon doing a nslookup to a different DNS server, ie. nslookup www.voip-info.org 141.1.1.1 I get back 64.65.102.50. It seems that your DNS configuration might be broken. I hear you and so does Jim, the owner of the system, the domain and the Wiki. I guess he is trying to solve this a.s.a.p. So far, he's been very reactive. I'm just filling his system with information on his invitation earlier on this list. :-) Thank you, Jim, for providing such a valuable resource to the Asterisk community! I hope we can sort these problems out. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] High Availability and Mass Deployment for Asterisk
Robert Hajime Lanning wrote: quote who=WipeOut Sharing the config files is the smallest problem.. its sharing SIP session and reistration information that is more of an issue.. And managing the data flows.. I wouldn't really worry about that. What happens when current PBX's fail. Does another PBX automaticaly come up and take over the PRI lines? The current expectation is that on PBX failure, current communications are broken and need to be re-established when the system comes up. If you are deploying for VoIP only, then network sync and failover is sort-of do-able. But, if it is being deployed with a hybrid (TDM and VoIP), then it gets hairy. I would expect that failover of the TDM side (without interuption) would be the hardest and most costly to implement. I would have a hot standby that gets syncs of dynamic data (voicemail, config changes...). On primary failure, the hot standby gets the primary's IP and someone gets to move the PRI (or other TDM connections) lines. It is kind of the same thing as if a channel bank failed. You are right in what you are saying.. I was thinking back to the original message that started this thread that talked about load balancing VoIP clients accross multiple servers.. Thats where my comments came from.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi Dan, Another problem I am seeing is I cannot delete any phone book entrys. This is very strange... Someone else with this issue? I cannot reproduce it here Just tried to delete Entry 12. Same problem here. And afterwards I can't start DIAX anymore, except by manually editing diax.cfg and adding a new blank entry at the end,ie. 12,, hth rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a bit frightened, guys
2. legal issues (what happens when an employee needs to call emergency personnel and the phone system doesn't work for whatever reason) I was worried about this until I realized that commercial systems were _no_ different in this regard. My current Meridian system is a PC in a fancy box with a UPS. The Toshiba and Panasonic and Siemens systems I've seen are no different. You're still dialing by wire in any PBX or KSU. I've tested the 911 capability of * and, using an extension trick given from the #asterisk IRC channel, dialling 911 just plays You will dial 911 in 5 seconds. If this was done in error, hang up now before actually zapping a trunk line (if all are busy) and dialling out. Hell my Meridian system doesn't even do that now! Regards. Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a bit frightened, guys
On Mon, 2003-11-03 at 11:27, Rich Adamson wrote: A couple of items to consider (in addition to the technical * implementation issues) are: Many thanks for your input, Rich - fortunately many of these issues don't pertain to our own environment, but they're still highly worth pointing out, and I'll certainly be mentioning them in wednesday's management meeting :) (now with 33% extra IT input!) Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Absolute Minimum Installation Packages
Hello, On Fri, 31 Oct 2003 10:24:32 -0600, David Gomillion wrote I can understand the size concerns for putting it in an appliance or what-not. However, my opinion is that, due to the low cost of hard disk space, it is cheaper for the company to go out and buy another hard disk to replace the extra 500 MB they wasted on a sub-optimal installation than to pay me to try to get the installation as small as possible. you're absolutely right here, for the cost of a 128mb cf card i can get a 40gb hdd, where the space is not a concern What are the benefits to a really tiny installation, aside from possible appliance applications? Moreover, won't you still need a sizable hard disk for voice prompts, voicemail messages, sound file to direct people to dial the correct extension, etc? what i thought about was a closed box with some web interface that could serve as a voip gateway (and possibly as, say, web proxy/cache held on tmpfs?) not being a full blown server (there's a difference between convincing people to put a 30x30 box somewhere and making them put a high-tech server with raid, streamers whatnots. having the system run from a read-only medium (like a cf card with a tmpfs overlay - see http://translucency.sourceforge.net though haven't tried it yet) removes the need for backups extended reliability (nothing changes and if the data is somehow lost, restoring it is trivial). furthermore, if the fs is on a solid state device (not a hdd or cd), there are no moving parts (except for a cpu/sys fan), improving hardware robustness and reducing noise level. as for voicemail, etc. you can put another hdd (capacity!) in there just for that or keep it in ram (speed+no moving parts+cheaper than cf and voicemail tends to have rather short life-time - or doesn't it?). if the hdd breaks or you get a blackout, oh well, you lose at most some voicemail. if i could fit an * distro in 20mb (seems reasonable if started from a floppy-distro), it leaves me 100mb for voice prompts, which should be enough. Again, I may be WAY off track, but one of the things I really like about * is that I can update it easily. Wouldn't you lose some of the beauty by putting it in an appliance? you can build asterisk on another machine and update it via, say, scp to your heart's desire Moreover, I HATE Nortel because they have a user-unfriendly interface, proprietary controls, non-standard connections, and the like. It seems to me that by appliance-izing we would be inviting the same abuses that the current systems enjoy. I could see it becoming an issue of open-source software on extremely proprietary hardware, meaning the user can modify their system if they can figure out how to get in it. what about ssh? the sshd isn't *this* heavy, is it? putting * in a closed box is appliance-izing it [nice word :)] in the eyes of the end-user (clicks here and there w/o all the *.conf voodoo), but leaves full power to the more competent users who can figure their way through ssh and asterisk's conf files Of course, all of this is in the assumption that the end- user wants to own their PBX. I know I do. I think that we should be focusing on a useful administrative interface, database-based extension definitions, and other features that will advance the power, flexibility, and usability of * instead of shrinking the distro as much as possible. i think we should aim both to scale up (like, 10k+ phone systems running *) and down (home pbx system with a fritz or x100p and zero initial knowledge required). btw, my shrinking of the * distro to a few dozen mb doesn't stand in the way of expanding your server farm, does it? What am I missing? I see many people much smarter than I am excited about this, so I am sure I simply failed to consider how it will revolutionize everything. Not that it'll revolutionize anything, it's simply opening another (however niche) market for *. Awaiting your enlightenment (preferably sans-flame), David Gomillion regards, grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi, - Original Message - From: Peer Oliver schmidt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 2:28 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Hi Dan, Another problem I am seeing is I cannot delete any phone book entrys. This is very strange... Someone else with this issue? I cannot reproduce it here Just tried to delete Entry 12. Same problem here. And afterwards I can't start DIAX anymore, except by manually editing diax.cfg and adding a new blank entry at the end,ie. 12,, You're right. Solved now. Check prerelease 0.9.1 later today. Thanks for your help, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside and outside through * is not possible at all. I know I can not be the only one who has tried to do this. Please any help would be greatly appreciated. You need to get Asterisk onto a public IP address.. Using the DMZ function on the router will not work.. If you search the archives you will see that it has been attempted many times.. I don't believe a public IP address is required in this case. I've not actually tried * on a Linksys DMZ, however it appears that Linksys is exposing all tcp udp ports and only doing basic NAT. If that impression is true, it should work. The reason is not in the IP but in the SIP headers.. they will be sent out from the Asterisk server with the internal IP address of the server, this means that when the SIP UA reads the SIP message and responds it will respond to the incorrect IP address.. I don't think that is what keeping the original poster's system from working. The issue is one extension is configured for canreinvite=no and the other is canreinvite=yes. One extension believes all RTP must be passed through * while the other is attempting to negotiate a phone-to-phone RTP session, thus dropping the audio. There may be some exceptions somewhere, but asterisk located behind a nat box can work and others have done it. But, it really requires a basic understanding of how the sip protocol does call setup, the functions implemented in the sip phones, and the ability to see what each box is doing in order to set acceptable perameters in each. One of the key issues in making it work is an understanding that sip phones (not asterisk) initiates the majority of all actions. By that I mean: 1. sip phones must register with * on udp 5060, which is simple layer-3 functions that can be handled by 99% of all nat products. 2. sip phone to sip phone calls can be handled in two ways: a. canreinvite=no (all rtp traffic passes through asterisk on rtp udp ports that can be specified and properly handled by nat boxes) b. canreinvite=yes (allowing the two sip phones to negotiate the rtp channel without asterisk involvement) 3. In both 2a and 2b (for the original poster), the sip phones initiate the rtp negoitiation process and therefor asterisk does not have to rewrite the sip headers (only the sip phones). Asterisk already knows what the Internet address is of the remote sip phone because the sip phone told it (via it rewriting the header). The original poster should be able to get either 2a or 2b to work with the appropriate nat box mappings and sip configuration parameters. He can't expect it to work when you tell one sip phone to rtp one way and tell the second sip phone to do it different way. If the same original poster had indicated that 100 sip phones existed on the Internet and another 100 existed on his internal nat'ed network, then the answer to his question may be completely different. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
WARNING!!! (from bilbo.inter.net.il) The following message attachments were flagged by the antivirus scanner: Attachment [2.1] , scan failed: Internal error (0x11). Action taken: incomplete scan My asterisk server is inside my LAN. Our branch office is connected to here via VPN tunnel, traversing several FWs and VPN appliances. And we've been able to make sip to sip phone calls. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Monday, November 03, 2003 1:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing Shoval Tom wrote: Isn't putting asterisk on the public IP network a bad idea? Is it a bad idea?, Not really if you take the right precautions..From how you described your setup you have connected your server directly to the internet anyway.. If you nominated you Asterisk box as the DMZ host in your router it effectively is directly on the internet.. if you havent secured the box itself I suggest you do.. :) What about security? This is somthing that you will need to take care of.. Of course some people's opinions on securing a PC is to not connect it to the internet at all, of course that is a little silly.. You will have to decied on the level of security you are happy with.. This is a topic that can be debated for days so I will not get into it any further than that.. And how will all us newbies make the linux box as secure as possible? The quickest way is to setup an IPTABLES firewall.. You will need ports 5060 and 1 to 2 open for a default Asterisk install using SIP only.. (NOTE: make sure you know how to activate and deactivate IPTABLES from a command line because while you are playing there is a good chance you will lock yourself out of the server from any remote PC and you can even break Xwindows running locally with a firewall..) Later.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Monday, November 03, 2003 11:05 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing Robert Mann wrote: Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside and outside through * is not possible at all. I know I can not be the only one who has tried to do this. Please any help would be greatly appreciated. Robert, You need to get Asterisk onto a public IP address.. Using the DMZ function on the router will not work.. If you search the archives you will see that it has been attempted many times.. The reason is not in the IP but in the SIP headers.. they will be sent out from the Asterisk server with the internal IP address of the server, this means that when the SIP UA reads the SIP message and responds it will respond to the incorrect IP address.. So the basic rules where NAT is involved are.. Asterisk server must always be on a public IP address.. SIP UA's can be behind NAT but need nat=yes, canreinvite=no and qualify=yes set in the phone configuration in sip.conf.. Hope that helps.. 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] recording files for menues
That is correct. I'm able to get to your site using the IP address provided below. Since I get the same address (192.168.168.3) from four different ISPs (home, HQ, branch office, and dial-up to another one) I think it's safe to say your DNS configuration is what should be looked at first. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver schmidt Sent: Monday, November 03, 2003 1:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] recording files for menues Hello Olle, Olle, I can't reach the faq page, and haven't been able to for the last four days. I'm getting 504 gateway timeout errors. Gateway timeout indicates something with your web proxy ...or? I've been able to reach the Wiki all weekend, I've updated and created several pages... I also now that Jim have been working to speed things up, among them adding more SQL connections as we have had many hits at the same time when mailing a URL to the list... You should be able to reach it. Is it only this web site or do you get that error message somewhere else? I have not been able to reach www.voip-info.org as well using my default settings. Upon researching the problem I tried nslookup www.voip-info.org which returns an IP address of 192.168.168.3 which is obviously wrong. This is the answer from my local DNS server contacting the t-online DNS server. Upon doing a nslookup to a different DNS server, ie. nslookup www.voip-info.org 141.1.1.1 I get back 64.65.102.50. It seems that your DNS configuration might be broken. rgds pos ___ 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 hardphones? anyone?
On Mon, 2003-11-03 at 13:07, Roy Sigurd Karlsbakk wrote: hi all anyone that've heard of any working IAX hardphones yet? There is an unofficial firmware for the SNOM phones: http://www.marko.net/asterisk/archives/0208/0158.html although that thread seemed to go nowhere - may be worth chasing Mark about it :) http://www.awsystems.de/voip/snom100 also hints at it. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX hardphones? anyone?
On Mon, 2003-11-03 at 07:07, Roy Sigurd Karlsbakk wrote: hi all anyone that've heard of any working IAX hardphones yet? During Phreaknic, Mark showed the IAXY(sp?) a small maybe 2x3 board that was a single analog FXS port to IAX adapter. It was impressive that he was able to come into the con room, plug into the ethernet and without reconfiguring anything, he started making phone calls out of his office system. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gnophone problem
Dear Listers, I have the following problem: me and my father try to use 2 gnophones to talk to each other. We both registered at Iaxtel and both can call other numbers --- say FWD numbers, but when one of us tries to ring the other's gnophone, we get the the party you are trying to call is currently unregsitered or unavailable message and nothing happens on the other side. In the gnophone telephone settings dialog use asterisk is checked, and everything is filled in properly (I think), the peer and secret fields are left empty. Any help would be appreciated: TIA Andras ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Quick Question
I'm using * under RH9... When I go into production, I'll probably be changing distros, though. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Sussman Sent: Saturday, November 01, 2003 7:16 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Quick Question Apologies if there is a cleanly written and searchable FAQ that I could be directed to. I have no problem to RTFM if I can find the FM... Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... BTW, where would I find a useful FM? David -- David J. Sussman, MBA email: [EMAIL PROTECTED] web:http://www.processdevelopmentgroup.com phone: 248-212-7293 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a bit frightened, guys
2. legal issues (what happens when an employee needs to call emergency personnel and the phone system doesn't work for whatever reason) I was worried about this until I realized that commercial systems were _no_ different in this regard. My current Meridian system is a PC in a fancy box with a UPS. The Toshiba and Panasonic and Siemens systems I've seen are no different. You're still dialing by wire in any PBX or KSU. I've tested the 911 capability of * and, using an extension trick given from the #asterisk IRC channel, dialling 911 just plays You will dial 911 in 5 seconds. If this was done in error, hang up now before actually zapping a trunk line (if all are busy) and dialling out. Hell my Meridian system doesn't even do that now! The key to your response, Andrew, is that you gave it thought and due diligence in your design and operation. Can you picture some poor linux guru that didn't and he's on the stand in front of a jury and an attorney that eat's road-kill for practice? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail servermail and fromstring
On Mon, 3 Nov 2003, Senad Jordanovic wrote: The voicemails servermail and fromstring variables should change default values when email voicemail notification gets received by user. I change it, but received mail still shows Asterisk PBX in place of fromstring. Anyone knows is there anything else needs changing? Did you reload after you made the change? dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk compliance with RFC 2617 (qop, nc and cnonce) - in relation to sipcall.co.uk
Hi All, I am attempting to setup Asterisk with sipcall.co.uk. They use Intertex kit to provide the SIP service. Unfortunately Asterisk cannot seem to authenticate against Intertex. Having provided SIP debug info the provider has informed me that Asterisk does not appear to support 'qop', 'nc' and 'cnonce' which are used to stop replay attacks. So, does Asterisk support 'qop', 'nc' and 'cnonce'? If not, is anyone working on support? Before anyone says code it yourself, I would if I could! Thanks, Nathan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail servermail and fromstring
Did you reload after you made the change? dave Yes, many times. BTW, The servermail variable works fine. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: QoS What to do?
On Mon, Nov 03, 2003 at 11:32:11AM +0100, Roy Sigurd Karlsbakk wrote: If your DSL link is the bottleneck, rather than earlier hops back through the providers network, the provider could also prioritize VOIP packets going up the DSL line. That requires a cooperating provider, of course. You may also setup a linux box (or another QoS supporting router) on the inside and tune the communication with queueing there. Read the LARTC howto for more info. FIAIF at http://www.fiaif.net/ is an excellent piece of software which integrates firewall and QOS (based on lartc.org's wondershaper) functions. -- Poor girl looks as confused as a blind lesbian in a fish market. - Simon R. Green ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail servermail and fromstring
Senad Jordanovic wrote: The voicemails servermail and fromstring variables should change default values when email voicemail notification gets received by user. I change it, but received mail still shows Asterisk PBX in place of fromstring. Anyone knows is there anything else needs changing? AFAIK these only work with voicemail2.. check your extensions.conf and make sure you are using voicemail2 and not just voicemail.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where can i get the g.723 codec?
Hi all, can somebody tell me where i can get the g.723 codec for * ? Thanks. Regards, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail servermail and fromstring
AFAIK these only work with voicemail2.. check your extensions.conf and make sure you are using voicemail2 and not just voicemail.. Yap, that did it. :) Ta Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
On Mon, 2003-11-03 at 14:28, Thomas Haeger wrote: Hi all, can somebody tell me where i can get the g.723 codec for * ? http://store.yahoo.com/asteriskpbx/asteriskg729.html $10 per channel. I looked into the licensing costs for another product, and this is damn cheap. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hello all, I have a half working configuration: I have an asterisk box with one GS101 register to it in SIP mode and an IP10S in MGCP mode. I can dial IP10S from my GS101 and everything seems fine. But from my IP10S I can't dial any number (GS or anything else). All the version I use are the latest available Any Idea? Regards, Daniel Marian Danisek a écrit: rnc Info Lists wrote: Hi, -Original Message- The portion of extensions.conf is: exten = 3001,1,Dial(MGCP/aaln1,20) exten = 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20) Or aaln/1@ip should do just fine. However this doesn't explain why there is no dialtone on the phone.. Oh, one thought: Did you set your toneconfiguration to Europe or US ? If you choose custom you need to configure it another way... Florian Update: I changed the tone config to USA to match Asterisk. No change. I did notice that when I booted up everythign tonight that the MGCP SHOW ENDPOINTS now shows: Gateway 'ip10' at 0.0.0.0 (Dynamic) -- 'aaln/[EMAIL PROTECTED] in 'from-sip' is idle In the messages at start up there is: == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) -- MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate [chan_iax2.so]NOTICE[163851]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '192.168.0.5' (and thus its endpoint 'aaln/1') does not exist -- Setting hookstate of aaln/[EMAIL PROTECTED] to ONHOOK MGCP DEBUG shows the below lines repeating every couple of seconds: from 192.168.0.5:2427MGCP read: RSIP 1375 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1375', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Still no dialtone and not able to send or receive calls. Evidently there is a problem finding the phone. I can ping it from the Asterisk server so isn't a raw IP issue. On the phone there is the message Waiting for call manager Additional ideas are appreciated. Will keep plugging away at it. in sending you my mgcp.conf file, my ip10s mostly working fine... regards Marian ---mgcp.conf- [general] port = 2427 bindaddr = 192.168.1.253 [192.168.1.92] threewaycalling=yes transfer=yes callwaiting=yes callwaitingcallerid=yes host=192.168.1.92 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = John 92 line = aaln/1 [192.168.1.91] threewaycalling=yes transfer=yes callwaiting=no callwaitingcallerid=no host=192.168.1.91 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = Mary 91 line = aaln/1 Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Where can i get the g.723 codec?
This is the g.729 codec, but i want the g.723 -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Gavin Hamill Gesendet: Montag, 3. November 2003 15:44 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Where can i get the g.723 codec? On Mon, 2003-11-03 at 14:28, Thomas Haeger wrote: Hi all, can somebody tell me where i can get the g.723 codec for * ? http://store.yahoo.com/asteriskpbx/asteriskg729.html $10 per channel. I looked into the licensing costs for another product, and this is damn cheap. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a bit frightened, guys
Andrew Kohlsmith wrote: tested the 911 capability of * and, using an extension trick given from the #asterisk IRC channel, dialling 911 just plays You will dial 911 in 5 seconds. If this was done in error, hang up now before actually zapping a trunk line (if all are busy) and dialling out. Hell my Meridian system doesn't even do that now! Could you please share that extension logic with us, so we can document it for others? /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail servermail and fromstring
I am new to this so smack me if I am wrong but shouldn't it be serveremail not servermail? Maybe serveremail being wrong causes the fromstring not to function and the default * is using just happens to be the same thing your serveremail is set to. Robert - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 5:29 AM Subject: [Asterisk-Users] Voicemail servermail and fromstring The voicemails servermail and fromstring variables should change default values when email voicemail notification gets received by user. I change it, but received mail still shows Asterisk PBX in place of fromstring. Anyone knows is there anything else needs changing? ___ 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: AW: [Asterisk-Users] Where can i get the g.723 codec?
On Mon, 2003-11-03 at 14:51, Thomas Haeger wrote: This is the g.729 codec, but i want the g.723 Apologies - was too quick to jump :) I'm not aware of there being any G.723.1 codec pre-licensed for use with Asterisk. The code won't be hard to find, and will probably be publically available, but will need crazy $$$ licensing if you implement it in a production system... Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi , I even think to avoid using an installer mainly because the installer part is bigger that the application himself. What do you think? Dan, I agree that if an installer or registry entries are not needed then it makes an automated rollout much easier. Also makes it possible to run the program from a diskette/CD so as to be really portable between systems. However, the installer will be necessary for the acceptance by the non-geeks. I only had a short time to run your program last night but it worked well. Configuration was easy and it worked the first time! The problem with changing address book entries was encountered but that has already been reported. Will do more extensive testing tonight with the version from today. Thanks for a good program. Looking forward to it being GPL and the further development. Robert Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Ji, -Original Message- I have an asterisk box with one GS101 register to it in SIP mode and an IP10S in MGCP mode. I can dial IP10S from my GS101 and everything seems fine. But from my IP10S I can't dial any number (GS or anything else). Is the callmanager setting on the IP10S correct ? (i.e. pointing to the asterisk box) Can you show 'mgcp debug' output ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html As you can see they want a LOT of money. This is why I doubt there will ever be G.723.1 codec available for Asterisk. --Eric On Mon, 2003-11-03 at 08:28, Thomas Haeger wrote: Hi all, can somebody tell me where i can get the g.723 codec for * ? Thanks. Regards, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] a bit frightened, guys
Can you post the trick, as far as the zapping a channel and what not? That's something I've been looking for... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Monday, November 03, 2003 6:25 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] a bit frightened, guys 2. legal issues (what happens when an employee needs to call emergency personnel and the phone system doesn't work for whatever reason) I was worried about this until I realized that commercial systems were _no_ different in this regard. My current Meridian system is a PC in a fancy box with a UPS. The Toshiba and Panasonic and Siemens systems I've seen are no different. You're still dialing by wire in any PBX or KSU. I've tested the 911 capability of * and, using an extension trick given from the #asterisk IRC channel, dialling 911 just plays You will dial 911 in 5 seconds. If this was done in error, hang up now before actually zapping a trunk line (if all are busy) and dialling out. Hell my Meridian system doesn't even do that now! Regards. Andrew ___ 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] Where can i get the g.723 codec?
On Mon, 2003-11-03 at 15:14, Eric Wieling wrote: Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html From what I remember when I looked into this about a year ago, this isn't even the end of it, since whilst DSPG represent /most/ of the IP holders on the codec, there are still others, and if you want to be completely sure of being legally in the clear, then you must reach seperate licensing arrangements with them If only some of the hard-phones would use Speex or similar, then all these problems would Go Away, and the production costs for the phones could drop, giving the manufr. the same amount of margin, but at a lower market cost. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Good system board to use with TE410P?
On Sun, 2003-11-02 at 06:43, Scott Stingel wrote: Can anyone please tell me their experiences with the Tyan i7501 series (Xeon-basd), or recommend an alternate motherboard? I'm using a TE410P card in a Tyan S2721 motherboard (a.k.a Thunder i7500 Pro). I've had no problems whatsoever with the motherboard. It's works great. (And yes, I'm using it in a 2U rack-mount case.) Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi Robert, - Original Message - From: rnc Info Lists [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 5:08 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Hi , I even think to avoid using an installer mainly because the installer part is bigger that the application himself. What do you think? Dan, I agree that if an installer or registry entries are not needed then it makes an automated rollout much easier. Also makes it possible to run the program from a diskette/CD so as to be really portable between systems. However, the installer will be necessary for the acceptance by the non-geeks. There is no more an installer in the prerelease which I will post on my site later today. As all you must do is to put all the files from the archive in the same directory and run the executable from there. No other files in other directories or registry settings. If you have a diskette or a pocket drive, taking it out you take the whole application with you, including all the config files or call lists. I only had a short time to run your program last night but it worked well. Run the new prerelease later today ( I will post a message when it will be there). It has some improovments based on users requests and solve some detected bugs. Configuration was easy and it worked the first time! The problem with changing address book entries was encountered but that has already been reported. Solved in the new prerelease. Will do more extensive testing tonight with the version from today. Thanks for a good program. Looking forward to it being GPL and the further development. Thanks a lot for your feedback, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
The makers of hardphones prolly get their G72x licensing by using a DSP that already has a license. The DSP can't be that expensive. I wish someone would make a PCI card with something like 8 of these chips on it and sell it cheap. Should be pretty easy to build a codec for Asterisk that uses the DSP card. On Mon, 2003-11-03 at 09:39, Gavin Hamill wrote: On Mon, 2003-11-03 at 15:14, Eric Wieling wrote: Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html From what I remember when I looked into this about a year ago, this isn't even the end of it, since whilst DSPG represent /most/ of the IP holders on the codec, there are still others, and if you want to be completely sure of being legally in the clear, then you must reach seperate licensing arrangements with them If only some of the hard-phones would use Speex or similar, then all these problems would Go Away, and the production costs for the phones could drop, giving the manufr. the same amount of margin, but at a lower market cost. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, Florian Overkamp a crit: Ji, -Original Message- I have an asterisk box with one GS101 register to it in SIP mode and an IP10S in MGCP mode. I can dial IP10S from my GS101 and everything seems fine. But from my IP10S I can't dial any number (GS or anything else). Is the callmanager setting on the IP10S correct ? (i.e. pointing to the asterisk box) Yes it is Can you show 'mgcp debug' output ? I have attached the debug trace from dialling extension 326 Regards, Daniel Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com MGCP read: NTFY 6611 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 X: 6746d764 O: hd from 192.168.10.10:2427Verb: 'NTFY', Identifier: '6611', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 6611 OK to 192.168.10.10:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- Creating connection for aaln/[EMAIL PROTECTED] in cxmode: sendrecv callid: 414339df6746d764 We're at 192.168.10.254 port 17648 Answering with capability 4 Posting Request: CRCX 8 aaln/[EMAIL PROTECTED] MGCP 1.0 C: 414339df6746d764 L: p:20, a:PCMU M: sendrecv X: 6746d764 v=0 o=root 31799 31799 IN IP4 192.168.10.254 s=session c=IN IP4 192.168.10.254 t=0 0 m=audio 17648 RTP/AVP 0 a=rtpmap:0 PCMU/8000 to 192.168.10.10:2427 -- MGCP Asked to indicate tone: dl on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 9 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 6746d764 R: hu(N), hf(N), D/[0-9#*](N) S: dl to 192.168.10.10:2427 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down MGCP read: 200 8 OK I: 8 v=0 o=- 8 0 IN IP4 192.168.10.10 s=- c=IN IP4 192.168.10.10 b=AS:81 t=0 0 a=sendrecv m=audio 3 RTP/AVP 0 a=ptime:20 from 192.168.10.10:2427Verb: '200', Identifier: '8', Endpoint: 'OK', Version: '(null)' 2 headers, 9 lines Capabilities: us - 4, them - 4, combined - 4 Non-codec capabilities: us - 1, them - 0, combined - 0 MGCP read: 200 9 OK from 192.168.10.10:2427Verb: '200', Identifier: '9', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: NTFY 6612 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 X: 6746d764 O: 3 from 192.168.10.10:2427Verb: 'NTFY', Identifier: '6612', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 6612 OK to 192.168.10.10:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '3' -- MGCP Asked to indicate tone: dl on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 10 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 6746d764 R: hu(N), hf(N), D/[0-9#*](N) S: dl to 192.168.10.10:2427 -- MGCP asked to indicate -1 'UNKNOWN' condition on channel MGCP/aaln/[EMAIL PROTECTED] -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 11 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 6746d764 R: hu(N), hf(N), D/[0-9#*](N) to 192.168.10.10:2427 -- MGCP mgcp_hangup(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED] -- Delete connection 8 aaln/[EMAIL PROTECTED] with new mode: sendrecv on callid: 414339df6746d764 Posting Request: DLCX 12 aaln/[EMAIL PROTECTED] MGCP 1.0 C: 414339df6746d764 X: 6746d764 I: 8 to 192.168.10.10:2427 -- MGCP Asked to indicate tone: ro on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 13 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 6746d764 R: hu(N), hf(N), D/[0-9#*](N) S: ro to 192.168.10.10:2427 MGCP read: 200 10 OK from 192.168.10.10:2427Verb: '200', Identifier: '10', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: 200 11 OK from 192.168.10.10:2427Verb: '200', Identifier: '11', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: 250 12 OK P: PS=21,OS=3612,PR=0,OR=0,PL=0,JI=0,LA=0 from 192.168.10.10:2427Verb: '250', Identifier: '12', Endpoint: 'OK', Version: '(null)' 2 headers, 0 lines MGCP read: 200 13 OK from 192.168.10.10:2427Verb: '200', Identifier: '13', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: NTFY 6613 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 X: 6746d764 O: 2 from 192.168.10.10:2427Verb: 'NTFY', Identifier: '6613', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 6613 OK to 192.168.10.10:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '2' -- MGCP Asked to indicate tone: ro on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 14 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 6746d764 R: hu(N), hf(N), D/[0-9#*](N) S: ro to 192.168.10.10:2427 MGCP read: 200 14 OK from 192.168.10.10:2427Verb: '200', Identifier:
[Asterisk-Users] Aastra 480 ADSI keypad problem
This is a really cool phone, except one problem, searched the archives and this was brought up before. Just wondering if anyone figured out how to solve it. I'm having the same problem as these previous posts... --- posted 06/09/03 Whenever I try using the voicemail through my ADSI display, it disables my # buttons. If I hit listen through the ADSI display, I can not delete messages. The 7 button no longer does anything... --- posted 06/09/03 I have the same problem. I use an Aastra 480 phone and as long as I don't touch any of the ADSI soft-buttons then my keypad stays active and the downloaded script works great. But as soon as I hit listen through the ADSI display, all of my normal 0-9*# keys get disabled and the script no longer maps any more options to my soft buttons. --- Any suggestions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OHT in fxs hates my answering machine + self fix
I have had this problem for a while where my fxs device has an answering machine on it and getting a call will hang up for no reason. I timed how long it took to hang up on me , 6 seconds , so I greped 6000 in zaptel src and found some code about OHT which was not present before sept back from before I had this issue. I patched the OHTcode by adding #ifdef USE_OHT #endif around the new code and that fixed me. question is will this hurt me at all and perhaps can future versions have a backwards version of this in the makefile like #ifndef and -D_DISABLE_OHT This is the kind of info I long to see in this list so I feel obligated to post it. Do you Yahoo!? Exclusive Video Premiere - Britney Spears
Re: [Asterisk-Users] Where can i get the g.723 codec?
Hi Thomas, Unless you have a *very* specific need to use G.723.1 for compatibility with someone else, forget it. It is pretty much an obsolete product. Licencing is also a pain, as there is not patent pool for it. G.729 is expensive to licence, but at least it is relatively strightforward. If you think you will save some bits using G.723.1 instead of G.729, think again. The saving is minute, because of the huge overheads IP imposes. Regards, Steve Thomas Haeger wrote: Hi all, can somebody tell me where i can get the g.723 codec for * ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /var/spool/asterisk/outgoing
Lists wrote: I am having a wired issues with the outgoing calls here is my queue file Channel: IAX2/[EMAIL PROTECTED]/NUM MaxRetries: 1 RetryTime: 600 WaitTime: 300 Context: playoutstart Extension: s Priority: 1 If someone picks up the phone, it works great, if it gets a voicemail, it plays the message, however it also appened this to the file: Retry: 1 (1067871901) and then it does it again. It will only do that if someone DOES not pick up the phone. Why? Thanks, Michael Eliminate the MaxRetries line.. What is it that you are trying to do with this?? It looks like you are calling a nufone number and then putting the call into your start manu.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
On Mon, 2003-11-03 at 15:47, Eric Wieling wrote: The makers of hardphones prolly get their G72x licensing by using a DSP that already has a license. The DSP can't be that expensive. I wish someone would make a PCI card with something like 8 of these chips on it and sell it cheap. Should be pretty easy to build a codec for Asterisk that uses the DSP card. Hm, interesting idea. I wonder if one of the clauses in the purchase of the DSP chips is that you must use them in a complete embedded device, rather than a general-purpose peripheral as you suggest QuickNet certainly did this with their Windows PhoneJack LineJack, but interestingly the Linux LineJack had the hardware DSP facility removed IIRC - I'm guessing the 'open'-ness of Linux just frightened the legal people :( Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
I must agree with Eric on this one. I did testing with g723.1 pass thru between two cisco ATA's and you can fit two calls in the same bandwidth as one g729 call. But without a codec in * its pretty much pointless. Also I have emailed these guys about the g723.1 lic they NEVER email back. Even with the crack headed g729 lic setup it still works. bkw On Mon, 3 Nov 2003, Eric Wieling wrote: Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html As you can see they want a LOT of money. This is why I doubt there will ever be G.723.1 codec available for Asterisk. --Eric On Mon, 2003-11-03 at 08:28, Thomas Haeger wrote: Hi all, can somebody tell me where i can get the g.723 codec for * ? Thanks. Regards, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Where can i get the g.723 codec?
Thanks Steve, there is no special reason for me for using g.723. I will take g.729. It seems to be easier :-) Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Steve Underwood Gesendet: Montag, 3. November 2003 17:14 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Where can i get the g.723 codec? Hi Thomas, Unless you have a *very* specific need to use G.723.1 for compatibility with someone else, forget it. It is pretty much an obsolete product. Licencing is also a pain, as there is not patent pool for it. G.729 is expensive to licence, but at least it is relatively strightforward. If you think you will save some bits using G.723.1 instead of G.729, think again. The saving is minute, because of the huge overheads IP imposes. Regards, Steve Thomas Haeger wrote: Hi all, can somebody tell me where i can get the g.723 codec for * ? ___ 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: IAX2 Java library (was Re: [Asterisk-Users] New IAX software phone (for WIndows platform))
Alastair Maw wrote: On 03/11/03 00:25, Mark Spencer wrote: As a side note, I strongly would like to see someone implement a client using libiax2 which implements IAX2 instead of the (now obsolescent) IAX version 1. I'm implementing a Java-based IVR server (and yes, I know Asterisk does IVR, and no, it's not flexible enough to do what I want and no, it doesn't integrate well with the Java systems we have, etc. hence my doing this). Are you mad? What is not flexable enough for you? Java knows what STDIN and STDOUT is, right? What more do you need? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compliance with RFC 2617 (qop, nc and cnonce) - in relation to sipcall.co.uk
On Mon, 2003-11-03 at 15:02, nathan wrote: Hi All, I am attempting to setup Asterisk with sipcall.co.uk. They use Intertex kit to provide the SIP service. Unfortunately Asterisk cannot seem to authenticate against Intertex. Having provided SIP debug info the provider has informed me that Asterisk does not appear to support 'qop', 'nc' and 'cnonce' which are used to stop replay attacks. Went up the same route myself, and got the same answers from Sipcall and Intertex. The only time I was ever able to connect to Sipcall, even with an Intertex modem in place at my end, was using MS Messenger. The question I now ask is why only Intertex based systems require this? Asterisk registers with other providers, FWD, sipphone, iptel, nikotel etc... Do they suffer replay attacks? -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
You can port forward the 5060 SIP port and use externip keyword in sip.conf to have it working behind a NAT. Martin On Mon, 3 Nov 2003, WipeOut wrote: Robert Mann wrote: Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside and outside through * is not possible at all. I know I can not be the only one who has tried to do this. Please any help would be greatly appreciated. Robert, You need to get Asterisk onto a public IP address.. Using the DMZ function on the router will not work.. If you search the archives you will see that it has been attempted many times.. The reason is not in the IP but in the SIP headers.. they will be sent out from the Asterisk server with the internal IP address of the server, this means that when the SIP UA reads the SIP message and responds it will respond to the incorrect IP address.. So the basic rules where NAT is involved are.. Asterisk server must always be on a public IP address.. SIP UA's can be behind NAT but need nat=yes, canreinvite=no and qualify=yes set in the phone configuration in sip.conf.. Hope that helps.. 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] Where can i get the g.723 codec?
Eric Wieling wrote: The makers of hardphones prolly get their G72x licensing by using a DSP that already has a license. The DSP can't be that expensive. I wish someone would make a PCI card with something like 8 of these chips on it and sell it cheap. Should be pretty easy to build a codec for Asterisk that uses the DSP card. So you want to deal with the latency involved in two PCI bus transits?Asterisk doesn't need a old skewl DSP. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
QuickNet certainly did this with their Windows PhoneJack LineJack, but interestingly the Linux LineJack had the hardware DSP facility removed IIRC - I'm guessing the 'open'-ness of Linux just frightened the legal people :( IIRC the DSP is still enabled in Linux; it's just the g.729a codec that's gone. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
Martin Pycko wrote: You can port forward the 5060 SIP port and use externip keyword in sip.conf to have it working behind a NAT. Martin Martin, Is externip and new parameter?? Does it do a similar thing for the server as what nat=yes does for the phone? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P troubles
Maybe you need the straight through cable. Martin On Mon, 3 Nov 2003 [EMAIL PROTECTED] wrote: Hi, At least I have one E1 to test my E100P. My telco company in Spain has installed one LiteSpan 1540 NT (UTR 2M) I make a crossover cable between E100P and UTR. 1 - 4 2 - 5 after loading drivers red led on e100p is blinking and alarm is flashing on UTR. What is wrong ? my zaptel.conf inf: span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 best regards, Jorge Castellet [EMAIL PROTECTED] - Original Message - From: rnc Info Lists [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 4:08 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Hi , I even think to avoid using an installer mainly because the installer part is bigger that the application himself. What do you think? Dan, I agree that if an installer or registry entries are not needed then it makes an automated rollout much easier. Also makes it possible to run the program from a diskette/CD so as to be really portable between systems. However, the installer will be necessary for the acceptance by the non-geeks. I only had a short time to run your program last night but it worked well. Configuration was easy and it worked the first time! The problem with changing address book entries was encountered but that has already been reported. Will do more extensive testing tonight with the version from today. Thanks for a good program. Looking forward to it being GPL and the further development. Robert Germany ___ 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] New IAX software phone (for WIndows platform)
And Visual Basic? Please. What precisely is the problem with it? Or are you just a language nazi? I don't like VB any more than you do but if the thing works, who cares what it was written in. Nobody's asking me to maintain it. Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
I would be happy to do so for the advantages of G723.1. i.e. great sounding calls at a very low bandwidth. I suspect that the cost of running the data over the PCI bus multiple times would be more than offset by the faster compression/decompression provided by the DSP. On Mon, 2003-11-03 at 10:41, Jeremy McNamara wrote: Eric Wieling wrote: The makers of hardphones prolly get their G72x licensing by using a DSP that already has a license. The DSP can't be that expensive. I wish someone would make a PCI card with something like 8 of these chips on it and sell it cheap. Should be pretty easy to build a codec for Asterisk that uses the DSP card. So you want to deal with the latency involved in two PCI bus transits?Asterisk doesn't need a old skewl DSP. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Jeremy McNamara wrote: If not registered, nothing works and the application closes by himself. This is a very bad behavior. You only need to register if you plan on receiving a call from Asterisk and your IP is dynamic or you need to punch thru a NAT/Firewall edge device. An error message would be more helpful, I have to agree. And Visual Basic? Please. What is wrong with Visual Basic? I always thought, it is the solution that counts, not the programming language. Am I missing something? *I* think it is great to have an IAX client available that kind of works out of the box. There are rough edges, but the start looks promising. -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
On Monday 03 November 2003 06:17, Dan wrote: P.S. I'll post later today a new prerelease (0.9.1) with some bug fixes and some users requested improovements. Keep on eye on this list! As this has become quite popular and is taking up a significant number of postings on this list, might I suggest that a new list be made for Windows clients? I don't mind the announcements (and I think, they're most welcome), but I think the support issues could best be handled on a separate list. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Subject: Re: [Asterisk-Users] Where can i get the g.723 codec?
I hope this doesn't show up twice (posted from wrong mail adr.) Hi Eric, You can actually get boards like this already from companies like Mapletree. Its a hardware pci carrier card where you add the number of DSP modules that you need. This hardware may be a bit 'high end' for most users on this list but several people seems to address this issue pretty often. One card may be equipped with dsp's to handle 488 simultanoeus sessions in any mix of supported codecs(incl. G.723.1 and G729..). I whish that I had the time to make the 'glue logic' thats needed to connect the * codes api with Mapletree busmastering codecs channels. P.S. I am NOT a Mapletree salesperson. -The makers of hardphones prolly get their G72x licensing by using a DSP -that already has a license. The DSP can't be that expensive. I wish -someone would make a PCI card with something like 8 of these chips on it -and sell it cheap. Should be pretty easy to build a codec for Asterisk -that uses the DSP card. -On Mon, 2003-11-03 at 09:39, Gavin Hamill wrote: On Mon, 2003-11-03 at 15:14, Eric Wieling wrote: Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html From what I remember when I looked into this about a year ago, this isn't even the end of it, since whilst DSPG represent /most/ of the IP holders on the codec, there are still others, and if you want to be completely sure of being legally in the clear, then you must reach seperate licensing arrangements with them If only some of the hard-phones would use Speex or similar, then all these problems would Go Away, and the production costs for the phones could drop, giving the manufr. the same amount of margin, but at a lower market cost. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
Let's keep this positive. Somebody took the time to try to make something useful. He's not charging for it. If you don't like it, don't use it. If you have a problem with VB, port it to C. My pair of pennies, David Gomillion -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy McNamara Sent: Monday, November 03, 2003 10:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Dan wrote: If not registered, nothing works and the application closes by himself. This is a very bad behavior. You only need to register if you plan on receiving a call from Asterisk and your IP is dynamic or you need to punch thru a NAT/Firewall edge device. And Visual Basic? Please. Jeremy McNamara ___ 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] Asterisk behind LinkSys NAT Routing
It's new. It prevents asterisk from putting the private IP in the messages that asterisk sends with SIP. Martin On Mon, 3 Nov 2003, WipeOut wrote: Martin Pycko wrote: You can port forward the 5060 SIP port and use externip keyword in sip.conf to have it working behind a NAT. Martin Martin, Is externip and new parameter?? Does it do a similar thing for the server as what nat=yes does for the phone? 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
[Asterisk-Users] one way sound with x-lite (sip) -3rd attempt !
Hi List, Additional with the latest tries from the below I get a nice random seg fault when I hangup on PSTN. (With obviously no sound on x-lite, still!) asterisk -gc results after hanging up the pstn line in: -- Executing Hangup(SIP/1087997-d79f, ) in new stack == Spawn extension (sip-phone-out, h, 2) exited non-zero on 'SIP/phonenumber-d79f' Segmentation fault Since there is no normal release cycle can somebody give us advise which asterisk/X-Lite/chan_capi versions work well together ? (date and time of CVS version) Thanks in adavnce, Thorsten --- Hi all, Still having the one way sound problem. Any suggestions how to hunt the problem down ? Regards, Thorsten --- Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26 chan_capi On the IP side: X-lite (build: 1084) Calling and get calls on PSTN from X-Lite is no problem. We only get sound from PSTN to X-lite. Never from X.-lite to PSTN. The soundmeter on X-lite shows activity ... (not muted, correct device...) When pressing numbers while having these silent calls in x-lite is playing DTMFs at the PSTN phone side. sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to allow=all [1*phonenumber*] type=friend username=NAME secret=testpass auth=md5 nat=no host=dynamic reinvite=no dtmfmode=inband callerid=Test *phonenumber* context=sip-phone-out Any suggestions ? Thanks, Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] --PRI-- * --PRI-- modem bank - problems
Gentlemen We are attempting to use * in a simple switching application: +- office lines | V LEC --PRI-- * --PRI-- modem bank (56k dialup modems) The problem is that (even with no office lines active) the modems have difficulty establishing a connection, the connection is slow (way too slow for 56k modems) and the connections are prone to dropping. We have tried the Dial 'd' and 'c' options, but they did not help. The modems work fine when the PRI from the LEC goes straight into the modem bank. Any ideas about what is wrong here? Gary ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Peer Oliver schmidt wrote: And Visual Basic? Please. What is wrong with Visual Basic? I always thought, it is the solution that counts, not the programming language. Am I missing something? 1) Bloat 2) Borgware 3) Try running it on Linux/*BSD Not only is a really good win32 iax2 solution needed, but if someone is going to take the time, why not use a language that has a chance of being cross platform. And no, i'm not talking about Java either. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi Jeremy, - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 6:38 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Dan wrote: If not registered, nothing works and the application closes by himself. This is a very bad behavior. You only need to register if you plan on receiving a call from Asterisk and your IP is dynamic or you need to punch thru a NAT/Firewall edge device. You're right. The reason for me was that I wanted to make the program to have the look and feel of a standard phone. I will make a change in the registration window in order to choose if you want or not to register with the server you provide. And Visual Basic? Please. Please be more specific... It is something wrong to use it? Or.. your are one of the people who think that everything must be done in C (on *nix only)? Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
David Gomillion wrote: Let's keep this positive. Somebody took the time to try to make something useful. He's not charging for it. If you don't like it, don't use it. If you have a problem with VB, port it to C. I don't plan on using it. I will use mine, which is created in wxWindows and C++ and will run on Winsucks, UN*X and Mac. Yes, someday it will get released, maybe even the code if people are nice. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi, - Original Message - From: Peer Oliver schmidt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 7:19 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Jeremy McNamara wrote: If not registered, nothing works and the application closes by himself. This is a very bad behavior. You only need to register if you plan on receiving a call from Asterisk and your IP is dynamic or you need to punch thru a NAT/Firewall edge device. An error message would be more helpful, I have to agree. Which do you think is the best way to provide this functionality. I intend to do it like that: - put a checkbox in the registration window (yes, will be one in the next release..;)) - if you check the box, then a register procedure sis triggered. - if not, it just use the info provided to compose the dial address together with the phone number. If you need to dial an IAX address containing name not numbers, then you must define a phonebook entry for that. It is ok for you? Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: IAX2 Java library (was Re: [Asterisk-Users] New IAX software phone (for WIndows platform))
On 03/11/03 16:35, Jeremy McNamara wrote: I'm implementing a Java-based IVR server (and yes, I know Asterisk does IVR, and no, it's not flexible enough to do what I want and no, it doesn't integrate well with the Java systems we have, etc. hence my doing this). Are you mad? What is not flexable enough for you? Java knows what STDIN and STDOUT is, right? What more do you need? Not wanting to start a flamewar, but... - I can't possibly fork a whole JVM process for each caller. It's much too inefficient. This needs to support hundreds of simultaneous calls, and the GNU Java compiler just isn't good enough for our needs. I guess I could write an AGI wrapper script which connected to the Java server over a TCP connection or something and piped the stdin/out down the line to it. - We'd like to use Java because: - Need to do RMI to existing systems. Can't be bothered with all the CORBA nonsense. - It's more maintainable within our organization. - We have lots of existing components to support. - It does all the interoperability stuff we need very nicely, so we save time once the system is built (XML, etc.). - We like it. :) - I need access to the raw audio streams in realtime for various reasons (need to do DSP stuff for some clients, etc). Can I get this easily with AGI? Along with this, I need to be able to play audio from a URL. I don't want to have to download the whole file from the URL in order to play it - it wants to be streamed. Is this possible with AGI? The docs aren't very good for AGI, so I don't really know... - I need to be able to generate large amounts of audio in realtime, conference people together but then only play an audio file to one person within the conference, etc. I don't think AGI is flexible enough to do this. - I'd like to be able to move from Asterisk to something else if I need to. This is why originally I was doing things using SIP/RTP. - The documentation for AGI is very poor. I know it is for IAX, too, but I can see a Java IAX library being useful for client development too, and I'd like to give a little back to the * community, you know? There are other reasons, but I haven't the time to explain right now. The above are the most important. -- Alastair Maw MX Telecom http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
At 12:11 PM 11/3/2003, you wrote: And Visual Basic? Please. What precisely is the problem with it? Or are you just a language nazi? I don't like VB any more than you do but if the thing works, who cares what it was written in. Nobody's asking me to maintain it. I was willing to give vb a chance at one time, but won't touch it any more. Speaking from firsthand experience here is what led to the eventual scrapping of a commercial product written in VB and a vow we will never use it again corporately. 1) 95 series / nt series stuff just behaved differently and no amount of property settings or traps changed that. 2) third party components are mostly supplied without source code. Vendors come and go like the wind so if you depend on something and it turns out to have a bug, often the only fix is a redo of the code. Alternatively you can write your own, but you pretty much have to do it with c to get the power to do what you need otherwise you would have just implemented in vb to start with. 3) There were many many bugs in the runtime libs - this may have changed in later versions (we tried 3, 4, and 5 with our code, which fixed some bugs and introduced new ones.) We finally just gave up on it since the customers using it were having constant issues we could not fix without replacing whole sections of VB with c code. 4) components changed behaviour with new releases of the compiler, and needed code fixes to accomdate, so trying later versions was somewhat of a one way street. 5) The RAD of visual c and its templating is very close to that in vb, but you get the component sources, and can fix them if need be. 6) if you need special pieces you are interfacing c to c which is much simpler. (remember vb uses a pascal style strings and C is an SZ - extra overhead and hair pulling to interface, especially when vb has several variants on pascal style length storage) you have to end up asking yourself, is it still vb after I replace all the parts with c to get it to do what I want ? if the answer is more than 50%, why start with vb in the first place ? Andrew ___ 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] Intel Performance Primitives
Hey all, For those of you who are really worried about asterisk performance, I thought I might alert you to a toy you might play around with. The Intel Performance Primitives contain a number of optimized functions for use in digital signal processing that could help with echo cancellation, codec transformations, etc. I don't have any idea how useful this would be in Real Life (actual performance gain, license compatibility, etc), but there you go... http://www.intel.com/software/products/ipp/ipp30/ --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Proper syntax for the Cut application?
Hi. I am looking for the proper syntax for the Cut application. I am
working on a Feature Code extension that drops a caller directly into
a voicemail box. Here is what I have:
exten = _55.,1,Answer()
exten = _55.,2,Cut(VMEXT=EXTEN|55|2)
exten = _55.,3,Voicemail(u${VMEXT})
exten = _55.,4,Hangup()
When I dial 551100, the system tries to process this but I get dropped
immediately. Here is the output:
-- Executing Answer(SIP/ppc-6aa2, ) in new stack
-- Executing Cut(SIP/ppc-6aa2, VMEXT=EXTEN|55|2) in new stack
-- Executing Voicemail(SIP/ppc-6aa2, u) in new stack
== Parsing '/etc/asterisk/voicemail.conf': Found
WARNING[1234379840]: File app_voicemail.c, Line 836 (leave_voicemail):
No entry in voicemail config file for ''
Obviously the new variable VMEXT is empty. Why is that? I read the
source for app_cut.c and the syntax looks correct. I am asking for the
second field which should be '1100'. I have tried several versions of
this, including changing the line from: exten =
_55.,2,Cut(VMEXT=EXTEN|55|2) to: exten =
_55.,2,Cut(VMEXT=${EXTEN}|55|2)
I have also tried changing the parsing function by parsing as such:
exten = _55.,2,Cut(VMEXT=EXTEN|5|2).
None of these put ANYTHING into the ${VMEXT} variable.
Please help!
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Re: [Asterisk-Users] one way sound with x-lite (sip) -3rd attempt !
Thorsten Trapp wrote: Hi List, Additional with the latest tries from the below I get a nice random seg fault when I hangup on PSTN. (With obviously no sound on x-lite, still!) asterisk -gc results after hanging up the pstn line in: -- Executing Hangup(SIP/1087997-d79f, ) in new stack == Spawn extension (sip-phone-out, h, 2) exited non-zero on 'SIP/phonenumber-d79f' Segmentation fault Since there is no normal release cycle can somebody give us advise which asterisk/X-Lite/chan_capi versions work well together ? (date and time of CVS version) I am currently running.. X-Lite build 1082 (+ snom and GS phones) Asterisk CVS-10/01/03-09:50:16 fcpci-suse8.2-03.11.02 CAPI drivers chan_capi.0.2.5c On RedHat 9 with all the latest patches applied using a FritzPCI card.. It has been stable as a rock.. Hope that helps.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
Hi! I don't think that is what keeping the original poster's system from working. The issue is one extension is configured for canreinvite=no and the other is canreinvite=yes. One extension believes all RTP must be passed through * while the other is attempting to negotiate a phone-to-phone RTP session, thus dropping the audio. Are you sure this is 100% correct? I have some doubts since: - you'd have to consider all possible connection permutations between all clients and then set canreinvite= accordingly, which doesn't sound like it makes much sense - sip.conf is for * only, the data are not seen or read by the SIP UA themselves. Thus it would appear that it is up to * to permit/not permit a reinvite between the two UAs So bascially from my understanding things work like this: Once one of the SIP call parties has a canreinvite=no it won't matter what the other party's setting looks like, RTP traffic will travel through * anyway. Am I wrong here? Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail servermail and fromstring
Hi! The voicemails servermail and fromstring variables should change default values when email voicemail notification gets received by user. I change it, but received mail still shows Asterisk PBX in place of fromstring. Same here - please open a bug report on this. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Live real extensions.conf samples?
I consider good examples to be those of John Todd and Zac Sprackett, viz: http://www.loligo.com/asterisk/current/extensions.conf http://sprackett.com/asterisk/conf/extensions.conf If you lop the filename off each of those, you also get a directory of *all* their .conf files, also good reading. N.B.: In their respective sip.conf's and iax.conf's, while both of them change usernames and passwords to protect the innocent, IMHO, Todd does it in a way which leaves it clearer how to use those files. Good examples especially for the various commercial gateways out there. Hope this helps! -Chris - Original message - From: Ken Godee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Sun, 02 Nov 2003 15:35:28 -0700 Subject: [Asterisk-Users] Live real extensions.conf samples? It would be nice to see a real extensions.conf from a live business operation, every extensions.conf I've seen posted or been able to dig up so far would fail bad in a live business operation. I just have the beginings of mine and would like to make sure I don't miss anything. Most extensions.conf files I've seen wouldn't even let you dial 911 in thier dialplan. That's just something you don't want to forget! Not to mention that a business type extensions.conf needs to have several class of restrictions for different departments/people, most just have everything available to everyone, this is just not so in the real world. Not it mine anyway. If someone doesn't want to post you can alway email me direct. Thanks ___ 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] High Availability and Mass Deployment for Asterisk
quote who=WipeOut You are right in what you are saying.. I was thinking back to the original message that started this thread that talked about load balancing VoIP clients accross multiple servers.. Thats where my comments came from.. :) My goof, for not reading the start of the thread. That is even tougher still. Also, using a shared resource (VoIP access) to access a none shared resource (PSTN access), will be very hard. You would almost need to put the PSTN access on a third node, that is not clustered. Then when a call is made (any call, to PSTN or otherwise) would be locked to a node. Think of the full duration of a call as a transaction. This would have to be buried deap in Asterisk and the kernel. The kernel to do the network side of load balancing. And, Asterisk to keep state. Also, Asterisk would need to know which node is primary for which session. Asterisk will need to not do anything with a VoIP session that it is not the primary node for, unless it is taking over that session on behalf of a non-operational node. For this complexity, you are now really asking for the phones to fail. I think the best and most stable way is to partition your dialplan and assign X VoIP clients per Asterisk server. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
Hi! How to use that externip new parameter? Where in sip.conf and what is the format? thanks - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 3:34 PM Subject: Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing It's new. It prevents asterisk from putting the private IP in the messages that asterisk sends with SIP. Martin On Mon, 3 Nov 2003, WipeOut wrote: Martin Pycko wrote: You can port forward the 5060 SIP port and use externip keyword in sip.conf to have it working behind a NAT. Martin Martin, Is externip and new parameter?? Does it do a similar thing for the server as what nat=yes does for the phone? 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
