Re: [asterisk-users] Matching *, + and # in the dialplan
Tilghman Lesher [EMAIL PROTECTED] writes: exten = [0-9*#+].,... If that does not work, that is a bug and needs to be reported as such. Sadly that matches *james and 9foo... It would be nice if you could use normal regexes (e.g. with the pcre library) in extensions.conf. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Trunking
Brent Davidson [EMAIL PROTECTED] writes: I have several branch offices, each with their own Asterisk server (version 1.4.22.1) handling their PBX functions. All of these offices need to talk to each other. In sip.conf I created a peer entry for each office with a username of branch-user and a friend entry for every branch-user with the username being just the branch, for example: You should only need peer entries... type=user is dying. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got event 17 (Polarity Reversal)...
If by default Asterisk ignores all polarity events, then why does it cause the Dialplan to start? I did set answeronplarityswitch to no, however, I have had the problem occur once already, so, you suspicion might be correct. Jim Tzafrir Cohen wrote: On Fri, Oct 10, 2008 at 05:51:29PM -0400, Jim Duda wrote: Tzafrir, Thanks for the tip. I'm researching answeronpoliaryswitch. I suspect this will solve my issue. I never would have know to look for this. Thanks much! You made my day :-) Hmm... I might have misled you. By default Asterisk ignores all polarity events. Using the polarity events can be a useful feature, but I suspect that it is not the cause of your original problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about echo cancelation
2008/10/10 Eric ManxPower Wieling [EMAIL PROTECTED] All calls with a 2-wire analog piece have echo. You cannot perceive the echo because it happens so fast on non-VoIP connections. On VoIP calls you have significant extra latency while causes you you to perceive the echo. Do you mean generated locally or generated distantly ? I understand that VoIP extra latency sometrimes renders perceivable what was unperceivable before. What suprises me is to hear that media getways filter one-way only : as 2-wires analog devices produce echo, and every phone has 2-wires analog audio, in every call you've got at least 2 sources of echo : one in each endpoint. Echo must be removed before the call is converted to VoIP -- in your case the Media Gateway is the device that must remove echo. So, if Alice is hearing its own voice, 1. where does it most probably come from ? 2. where should it be removed ? For both, I would reply : 1. it most probably comes from Bob's phone (as other devices in-between are digital so voice can't leak from there), 2. Alice voice echo should canceled at every location: Bob's PBX, PSTN network (ISDN in the case I had in mind) and Alice's Media gateway Do you agree ? Olivier wrote: Hi, I'm using the following setup : Alice IPPhone --LAN- Media gateway PSTN --- Phone Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably, when users complains about earing their own voice, that means that distant phone or nearby equipment is leaking : Bob's phone is sending Alice's voice signal back to Alice. So, to properly cancel, I would say Media gateway should substract from incoming signal the signal that left the media gateway few ms before. Discussing here and there, some say that Media Gateway never work this way : it would only filters out locally generated echo. Do you agree with that ? If positive, then what can you do, if Bob's phone generate much echo ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is there a way
Steve Totaro wrote: My only wish is that Linux had a facility like XP to bridge NICs without running all sorts of commands for brctl. Just a GUI like XP. Last time I setup a bridge in Linux, I had to change many kernel options and rebuild the entire kernel to get bridging working properly. With XP, you just select the NICS, right click and select add to bridge. For linux, I find that running firestarter, ICS/Firewall is fine, my end game is to get all of my traffic to go over an OpenVPN tunnel at my colo which is the default gateway over OpenVPN. Windows seems to have the easiest method of getting this done. I've taken to using Debian derivatives lately, so your YMMV, but maybe this is helpful to you... I haven't had to rebuild any recent kernels for bridging. I do have to apt-get bridge-utils, but that's a trivial thing I do on any box I install. I also typically apt-get other userspace stuff like vlan, nmap, tcpdump and wireshark, etc. I've been using the following type of code in /etc/network/interfaces to effect bridges. When I want to bridge a tap device with openvpn, I do something similar to establish a bridge at boot time with only one physical ether attached. Then I put the final brctl add into a script which is invoked via the up option line in the openvpn conf file. Then it's all automatic. I don't (yet) know how to do it on other distros. The following fragment is used to connect to a redundant pair of asterisks for failover: # bridge of two ethers for alternative paths to SIP clients auto eth1 iface eth1 inet static address 0.0.0.0 netmask 255.255.255.0 auto eth2 iface eth2 inet static address 0.0.0.0 netmask 255.255.255.0 auto sipbr0 iface sipbr0 inet static address 192.168.1.13 netmask 255.255.255.0 broadcast 192.168.1.255 network 192.168.1.0 bridge-ports eth1 eth2 smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk For Windows ?
hi for asterisk users, please any asterisk distribution (or Trixbox) for windows ? (except for the Asterisk Win32) bicose asterisk for Win32 have a Free (limited) PBX Manager and i have a problem with it: 1. the asterisk Windows Service is not installed by default 2. unable to connect to it (no responce from it) help me please! thanks___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a way to test SIP credentials without making a call?
Is there a SIP packet that a SIP client can send to Asterisk to confirm that the credentials entered by the user are correct, without placing a call? We'd like to test the credentials when the user enters them, rather than wait until they try to make their first call. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got event 17 (Polarity Reversal)...
I founded this behaviour in the past. When the CO provides reversal polarity and the FXO port is configured to ignore polarity events, then a reversal polarity could be detected as ringing if the hardware/software is not well designed or configured. So, if the CO provides polarity reversal, why not set answer and release supervision to yes? Jorge Mendoza Jim Duda wrote: If by default Asterisk ignores all polarity events, then why does it cause the Dialplan to start? I did set answeronplarityswitch to no, however, I have had the problem occur once already, so, you suspicion might be correct. Jim Tzafrir Cohen wrote: On Fri, Oct 10, 2008 at 05:51:29PM -0400, Jim Duda wrote: Tzafrir, Thanks for the tip. I'm researching answeronpoliaryswitch. I suspect this will solve my issue. I never would have know to look for this. Thanks much! You made my day :-) Hmm... I might have misled you. By default Asterisk ignores all polarity events. Using the polarity events can be a useful feature, but I suspect that it is not the cause of your original problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?
hi, (i am no sur): the user credential is tested during SIP Registration Step thanks and tel me if this is a error - Original Message - From: Eric Chamberlain [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 11, 2008 5:20 PM Subject: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call? Is there a SIP packet that a SIP client can send to Asterisk to confirm that the credentials entered by the user are correct, without placing a call? We'd like to test the credentials when the user enters them, rather than wait until they try to make their first call. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk For Windows ?
hi for asterisk users, please any asterisk distribution (or Trixbox) for windows ? (except for the Asterisk Win32) Why use Windows? If you want something free and easy to use, download a pre-built Asterisk Linux CD. You could try download a Trixbox .iso and give it a go. I'm sure they're are others too... Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about echo cancelation
Olivier wrote: 2008/10/10 Eric ManxPower Wieling [EMAIL PROTECTED] All calls with a 2-wire analog piece have echo. You cannot perceive the echo because it happens so fast on non-VoIP connections. On VoIP calls you have significant extra latency while causes you you to perceive the echo. Do you mean generated locally or generated distantly ? I understand that VoIP extra latency sometrimes renders perceivable what was unperceivable before. What suprises me is to hear that media getways filter one-way only : as 2-wires analog devices produce echo, and every phone has 2-wires analog audio, in every call you've got at least 2 sources of echo : one in each endpoint. Where did you hear that media gateways filter one-way only? Any 2-wire analog leg will be a source of echo. Many, many, many calls do not have a 2-wire leg. Think cell/mobile or endpoints with PRI or T-1. Echo must be removed before the call is converted to VoIP -- in your case the Media Gateway is the device that must remove echo. So, if Alice is hearing its own voice, 1. where does it most probably come from ? 2. where should it be removed ? For both, I would reply : 1. it most probably comes from Bob's phone (as other devices in-between are digital so voice can't leak from there), 2. Alice voice echo should canceled at every location: Bob's PBX, PSTN network (ISDN in the case I had in mind) and Alice's Media gateway If you (Alice) are hearing echo then the echo canceling can be done any time after it leaves Bob's 2-wire circuit but before the audio is converted to VoIP on your end. Telcos echo cancel cell/mobile phone calls (also a high latency path) and long distance calls, but almost never do EC on local calls. This is why you seldom get echo when calling a mobile phone or a long distance number -- you mostly get it on local calls. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?
I should have clarified, we're only making outbound calls, not inbound, so there is no registration. On Oct 11, 2008, at 9:27 AM, Meftah Tayeb wrote: hi, (i am no sur): the user credential is tested during SIP Registration Step thanks and tel me if this is a error - Original Message - From: Eric Chamberlain [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 11, 2008 5:20 PM Subject: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call? Is there a SIP packet that a SIP client can send to Asterisk to confirm that the credentials entered by the user are correct, without placing a call? We'd like to test the credentials when the user enters them, rather than wait until they try to make their first call. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got event 17 (Polarity Reversal)...
On Sat, 11 Oct 2008, Jorge Mendoza wrote: I founded this behaviour in the past. When the CO provides reversal polarity and the FXO port is configured to ignore polarity events, then a reversal polarity could be detected as ringing if the hardware/software is not well designed or configured. So, if the CO provides polarity reversal, why not set answer and release supervision to yes? We need the flexability to answer either way... Here in the UK the (BT) exchange will do a polarity reversal to signal incoming CLI - it then send the CLI, *then* sends the ring signals, so answering on polarity reversal would be wrong. They also do a random polarity reversal most nights too - some sort of automated line testing. Eg. from my home box: Oct 7 01:40:28 NOTICE[15581] chan_zap.c: Got event 17 (Polarity Reversal)... Oct 9 03:53:06 NOTICE[19904] chan_zap.c: Got event 17 (Polarity Reversal)... Note the times... Gordon Jorge Mendoza Jim Duda wrote: If by default Asterisk ignores all polarity events, then why does it cause the Dialplan to start? I did set answeronplarityswitch to no, however, I have had the problem occur once already, so, you suspicion might be correct. Jim Tzafrir Cohen wrote: On Fri, Oct 10, 2008 at 05:51:29PM -0400, Jim Duda wrote: Tzafrir, Thanks for the tip. I'm researching answeronpoliaryswitch. I suspect this will solve my issue. I never would have know to look for this. Thanks much! You made my day :-) Hmm... I might have misled you. By default Asterisk ignores all polarity events. Using the polarity events can be a useful feature, but I suspect that it is not the cause of your original problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk For Windows ?
On Sat, Oct 11, 2008 at 05:13:50PM +0100, Meftah Tayeb wrote: hi for asterisk users, please any asterisk distribution (or Trixbox) for windows ? (except for the Asterisk Win32) Trixbox is a complete linux distribution that includes Asterisk, among other software components. It might be possible to build Asterisk on win32 (probably only on cygwin). Better chances for it working in 1.6. Anybody tried to make a cygwin package of it? http://cygwin.com/setup.html#package_contents bicose asterisk for Win32 have a Free (limited) PBX Manager and i have a problem with it: 1. the asterisk Windows Service is not installed by default 2. unable to connect to it (no responce from it) IIRC asterisk-win32 does not include a way to rebuild the packaging. Hence you must ask them for support. Or reimplement that. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentialswithoutmaking a call?
then this is a error from me, thanks - Original Message - From: Eric Chamberlain [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 11, 2008 6:03 PM Subject: Re: [asterisk-users] Is there a way to test SIP credentialswithoutmaking a call? I should have clarified, we're only making outbound calls, not inbound, so there is no registration. On Oct 11, 2008, at 9:27 AM, Meftah Tayeb wrote: hi, (i am no sur): the user credential is tested during SIP Registration Step thanks and tel me if this is a error - Original Message - From: Eric Chamberlain [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 11, 2008 5:20 PM Subject: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call? Is there a SIP packet that a SIP client can send to Asterisk to confirm that the credentials entered by the user are correct, without placing a call? We'd like to test the credentials when the user enters them, rather than wait until they try to make their first call. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk For Windows ?
- Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, October 11, 2008 6:27 PM Subject: Re: [asterisk-users] Asterisk For Windows ? On Sat, Oct 11, 2008 at 05:13:50PM +0100, Meftah Tayeb wrote: hi for asterisk users, please any asterisk distribution (or Trixbox) for windows ? (except for the Asterisk Win32) Trixbox is a complete linux distribution that includes Asterisk, among other software components. It might be possible to build Asterisk on win32 (probably only on cygwin). Better chances for it working in 1.6. Anybody tried to make a cygwin package of it? http://cygwin.com/setup.html#package_contents thanks for tha but i dont have the pocibility to use Linux, bicose : i am blind and no reliable screen reader for linux (KDE or GNOME) is found then i am using scren reader for windows please if you found other solution or a cigwin Package for Asterisk Contact me bicose asterisk for Win32 have a Free (limited) PBX Manager and i have a problem with it: 1. the asterisk Windows Service is not installed by default 2. unable to connect to it (no responce from it) IIRC asterisk-win32 does not include a way to rebuild the packaging. Hence you must ask them for support. Or reimplement that. yes, i dont love Asterisk for Win32 is no reliable / scalable and include a commercialisation fitur (PBX Manager) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk For Windows ?
my friend i have a problem with linux accessibility i dont have (not found) a screen reader for Gnome or KDE this is the reason tha i use windows but linux is realy best / fast / easy thanks - Original Message - From: Chris Rowson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 11, 2008 5:33 PM Subject: Re: [asterisk-users] Asterisk For Windows ? hi for asterisk users, please any asterisk distribution (or Trixbox) for windows ? (except for the Asterisk Win32) Why use Windows? If you want something free and easy to use, download a pre-built Asterisk Linux CD. You could try download a Trixbox .iso and give it a go. I'm sure they're are others too... Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk For Windows ?
On Sat, Oct 11, 2008 at 06:43:47PM +0100, Meftah Tayeb wrote: my friend i have a problem with linux accessibility i dont have (not found) a screen reader for Gnome or KDE this is the reason tha i use windows but linux is realy best / fast / easy Isn't text-mode better for a screen reader? Just 25 lines on the screen. I know of quite a few blind people who use Linux. All major distributions support brile-tty (sp?) for the installaer. At least with some tweaks. And it is certainly well supported. And there's also Emacspeak. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1 second delay when connecting calls
Hello, We are using asterisk 1.6, sangoma pri card, and Cisco 7960 phones. When we make or receive calls there is a delay before voice is heard. Anyone have any ideas on where to start to debug or has anyone seen this before. We have played with settings on pri, asterisk, and phones with no change. Thanks for your help and ideas in advance. Neal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk For Windows ?
you buntu also has orca. mike On Oct 11, 2008, at 9:59 AM, Tzafrir Cohen wrote: On Sat, Oct 11, 2008 at 06:43:47PM +0100, Meftah Tayeb wrote: my friend i have a problem with linux accessibility i dont have (not found) a screen reader for Gnome or KDE this is the reason tha i use windows but linux is realy best / fast / easy Isn't text-mode better for a screen reader? Just 25 lines on the screen. I know of quite a few blind people who use Linux. All major distributions support brile-tty (sp?) for the installaer. At least with some tweaks. And it is certainly well supported. And there's also Emacspeak. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk For Windows ?
Meftah Tayeb wrote: my friend i have a problem with linux accessibility i dont have (not found) a screen reader for Gnome or KDE this is the reason tha i use windows but linux is realy best / fast / easy thanks A quick search using Google gave me http://live.gnome.org/Orca Sound isn't working right now on my workstation so I can't test it but it is installed by default on my CentOS 5 workstation. Rod -- - Original Message - From: Chris Rowson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 11, 2008 5:33 PM Subject: Re: [asterisk-users] Asterisk For Windows ? hi for asterisk users, please any asterisk distribution (or Trixbox) for windows ? (except for the Asterisk Win32) Why use Windows? If you want something free and easy to use, download a pre-built Asterisk Linux CD. You could try download a Trixbox .iso and give it a go. I'm sure they're are others too... Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.1 + openais
Hello,
I followed the steps by Russell*
http://www.venturevoip.com/news.php?rssid=1980*
and I got it working for publish_event only. As soon as I add
subscribe_event, Asterisk doesn't start and I just get the following
message:
*Oct 11 6:38:04.340485 [CLM ] nodeget: trying to find node *
I have no idea what's wrong. There is not very much information about this
issue.
Where are my conf, just in case. Any idea please bring it up.
[EMAIL PROTECTED]:~# cat /etc/ais/openais.conf
# Please read the openais.conf.5 manual page
totem {
version: 2
secauth: off
threads: 0
interface {
ringnumber: 0
bindnetaddr: 192.168.1.0
mcastaddr: 226.94.1.1
mcastport: 5405
}
}
logging {
to_stderr: yes
to_file: yes
logfile: /tmp/ais
debug: off
timestamp: on
}
amf {
mode: disabled
}
[EMAIL PROTECTED]:~# cat /etc/asterisk/ais.conf
[device_state]
type=event_channel
publish_event=device_state
subscribe_event=device_state
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Re: [asterisk-users] Question about echo cancelation
2008/10/11 Eric ManxPower Wieling [EMAIL PROTECTED] Olivier wrote: 2008/10/10 Eric ManxPower Wieling [EMAIL PROTECTED] All calls with a 2-wire analog piece have echo. You cannot perceive the echo because it happens so fast on non-VoIP connections. On VoIP calls you have significant extra latency while causes you you to perceive the echo. Do you mean generated locally or generated distantly ? I understand that VoIP extra latency sometrimes renders perceivable what was unperceivable before. What suprises me is to hear that media getways filter one-way only : as 2-wires analog devices produce echo, and every phone has 2-wires analog audio, in every call you've got at least 2 sources of echo : one in each endpoint. Where did you hear that media gateways filter one-way only? From a media gateway vendor (mentioning its own products capabilities). That's the main reason I opened this thread as it surprised me a bit ... Any 2-wire analog leg will be a source of echo. Many, many, many calls do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? Think cell/mobile or endpoints with PRI or T-1. Echo must be removed before the call is converted to VoIP -- in your case the Media Gateway is the device that must remove echo. So, if Alice is hearing its own voice, 1. where does it most probably come from ? 2. where should it be removed ? For both, I would reply : 1. it most probably comes from Bob's phone (as other devices in-between are digital so voice can't leak from there), 2. Alice voice echo should canceled at every location: Bob's PBX, PSTN network (ISDN in the case I had in mind) and Alice's Media gateway If you (Alice) are hearing echo then the echo canceling can be done any time after it leaves Bob's 2-wire circuit but before the audio is converted to VoIP on your end. Telcos echo cancel cell/mobile phone calls (also a high latency path) and long distance calls, but almost never do EC on local calls. This is why you seldom get echo when calling a mobile phone or a long distance number -- you mostly get it on local calls. That's what I thought after reading white papers here and there -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1 second delay when connecting calls
Try setting canreinvite=no in each of the device sections on a couple of phones, reload chan_sip.so and see if that fixes things. It has fixed the issue when I've tried it. [EMAIL PROTECTED] wrote: Hello, We are using asterisk 1.6, sangoma pri card, and Cisco 7960 phones. When we make or receive calls there is a delay before voice is heard. Anyone have any ideas on where to start to debug or has anyone seen this before. We have played with settings on pri, asterisk, and phones with no change. Thanks for your help and ideas in advance. Neal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
Thanks, Steve,
That's what I am unsure of. I don't know how to limit 1 call per trunk. If
that's an easy thing to setup, I'd love to see it.
On Fri, Oct 10, 2008 at 10:20 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Oh, I thought you had logic to count the calls on the trunk. You should
limit each trunk to one call. This is the primary reason besides the email
that basically said that customer support structure has been changed and
anything beyond the Demarc would not be supported, I the Demarc is simply
their boxen, so unless it is on their side, you will not get any helpful
support from Bandwidth, plus they CCed over 500 people by address instead of
setting up a group.
http://www.bandwidth.com/content/support/?page=standardSupport
I am with Junction and while a trunk is not unlimited as far as price for
usage, the amount of trunks is unlimited (or at least as unlimited as it can
be since nothing is really unlimited). They asked that I try not to go over
one call per second for any real duration, and that I not hammer one LATA do
to limited interconnects.
The other thing was Junctions was very easy to sign up with, great support,
and configuration was a breeze.
As for Bandwidth, I think they are solid but due to recent changes and the
fact that you must pay per channel, as well as the setup process, I decided
they were not for me.
I will take a second look at your sip.conf and extensions.conf later to see
if something jumps out at me. I suspect since you are setting up two
separate trunks with Bandwidth, you need to limit each trunk to one call,
rather than two.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 9:47 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
externip messes up DTMF detection, and by messes up I mean it doesn't
detect it at all. Setting nat=yes or nat=no didn't make a difference either.
When the trunks are in use, the calls are fine, no dropped audio. It only
happens when a 3rd call is made and there's no trunk available.
Thanks :)
On Fri, Oct 10, 2008 at 7:09 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
You need to configure your box for nat settings, externip and other
settings in sip.conf and set nat=yes instead of nat=no.
One way audio is almost always a NAT issue and those are two glaring
things that would cause problems.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hi Steve,
It's behind a NAT/Firewall but SIP translation is enabled and removing
it from behind the firewall did nothing, it still dropped calls. The calls
connect and everything works, but it dies when all trunks are in use and
someone else tries to call out. It seems like even though both channels are
in use, it tries to connect to the 2nd trunk and thus kills the audio.
Nothing strange came up in Wireshark or the firewall logs.
Thanks.
On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and
someone tries to dial out, they cause another call to get one-way audio
(the
caller hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that
tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out
on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband
because we were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;…irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix
it added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
Re: [asterisk-users] Question about echo cancelation
Handsets use a 4-wire connection. Handsets with the the volume turned up could cause a form of echo as the microphone picks up the ear piece audio (I call this acoustic echo). Everything I said applies to 2-wire caused echo. Other types of echo is fairly uncommon and cannot be solved by normal echo canceling systems. Most echo canceling systems I've seen (mostly tellabs) only cancel echo in one direction. I suspect all of Digium's EC systems only do echo canceling in one direction as well. Olivier wrote: 2008/10/11 Eric ManxPower Wieling [EMAIL PROTECTED] Olivier wrote: 2008/10/10 Eric ManxPower Wieling [EMAIL PROTECTED] All calls with a 2-wire analog piece have echo. You cannot perceive the echo because it happens so fast on non-VoIP connections. On VoIP calls you have significant extra latency while causes you you to perceive the echo. Do you mean generated locally or generated distantly ? I understand that VoIP extra latency sometrimes renders perceivable what was unperceivable before. What suprises me is to hear that media getways filter one-way only : as 2-wires analog devices produce echo, and every phone has 2-wires analog audio, in every call you've got at least 2 sources of echo : one in each endpoint. Where did you hear that media gateways filter one-way only? From a media gateway vendor (mentioning its own products capabilities). That's the main reason I opened this thread as it surprised me a bit ... Any 2-wire analog leg will be a source of echo. Many, many, many calls do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk For Windows ?
El sáb, 11-10-2008 a las 11:07 -0700, Roderick A. Anderson escribió: A quick search using Google gave me http://live.gnome.org/Orca Sound isn't working right now on my workstation so I can't test it but it is installed by default on my CentOS 5 workstation. I've installed it on my laptop running debian sid and works pretty good. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net SIP : [EMAIL PROTECTED] Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?
Eric Chamberlain wrote: I should have clarified, we're only making outbound calls, not inbound, so there is no registration. Is there a particular reason you /can't/ register? It would seem that registration would provide the functionality you require, even if you're only making outbound calls. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?
On Oct 11, 2008, at 1:41 PM, Rob Hillis wrote: Eric Chamberlain wrote: I should have clarified, we're only making outbound calls, not inbound, so there is no registration. Is there a particular reason you /can't/ register? It would seem that registration would provide the functionality you require, even if you're only making outbound calls. In the case of a server like Asterisk, wouldn't sending a register disrupt the flow of inbound calls until the UA that normally handles inbound calls re-registers? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cli commands missing
I just loaded a new asterisk install (1.4.19) and found that the sip, iax, and extentions commands are missing from the cli and are not listed in help either? Any idea what could have happened or where these commands may have gone? Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?
Eric Chamberlain wrote: Is there a particular reason you /can't/ register? It would seem that registration would provide the functionality you require, even if you're only making outbound calls. In the case of a server like Asterisk, wouldn't sending a register disrupt the flow of inbound calls until the UA that normally handles inbound calls re-registers? Are you using the same credentials as existing extensions to make calls from different extensions? That would seem to be a particularly bad idea. You should be configuring /one/ sip extension per SIP phone. Those extensions that handle outgoing calls only could be put in a different number range, or have a letter prefixed or suffixed to the extension, but you should /not/ be using one configured extension for two different purposes. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cli commands missing
On Sat, Oct 11, 2008 at 08:17:09PM -0700, Eric Fort wrote: I just loaded a new asterisk install (1.4.19) and found that the sip, iax, and extentions commands are missing from the cli and are not listed in help either? Any idea what could have happened or where these commands may have gone? It probably means that the respective modules (chan_sip.so, chan_iax2.so, etc.) were not loaded. Do you have autoload enabled in modules.conf? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cli commands missing
autoload is enabled but all the modules are not loaded. Why would this be? what should I look at? The cli output and modules.conf are below. Eric modules.conf: ; ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=yes noload=chan_oss.so noload=chan_alsa.so noload=chan_phone.so [global] --- here's what's loaded: test*CLI module show Module Description Use Count res_musiconhold.so Music On Hold Resource 0 res_speech.so Generic Speech Recognition API 0 res_features.soCall Features Resource 0 res_monitor.so Call Monitoring Resource 0 res_adsi.soADSI Resource0 res_smdi.soSimplified Message Desk Interface (SMDI) 0 res_crypto.so Cryptographic Digital Signatures 0 res_agi.so Asterisk Gateway Interface (AGI) 0 res_indications.so Indications Resource 0 app_sayunixtime.so Say time 0 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 func_strings.soString handling dialplan functions 0 pbx_realtime.soRealtime Switch 0 app_cdr.so Tell Asterisk to not maintain a CDR for 0 app_sms.so SMS/PSTN handler 0 codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 app_page.soPage Multiple Phones 0 format_vox.so Dialogic VOX (ADPCM) File Format 0 app_echo.soSimple Echo Application 0 app_channelredirect.so Channel Redirect 0 chan_usbradio.so usb Console Channel Driver 0 21 modules loaded - Eric On Sat, Oct 11, 2008 at 8:40 PM, Tzafrir Cohen [EMAIL PROTECTED]wrote: On Sat, Oct 11, 2008 at 08:17:09PM -0700, Eric Fort wrote: I just loaded a new asterisk install (1.4.19) and found that the sip, iax, and extentions commands are missing from the cli and are not listed in help either? Any idea what could have happened or where these commands may have gone? It probably means that the respective modules (chan_sip.so, chan_iax2.so, etc.) were not loaded. Do you have autoload enabled in modules.conf? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
