Re: [Asterisk-Users] HITBSecConf2006 - Malaysia: Call for Papers
On 03/11/06 19:24 Praburaajan said the following: Greetings from Hack in The Box -- We are pleased to announce that the Call for Paper (CfP) for HITBSecConf2006 - Malaysia is now open! Set to take place from September 18th - 21st 2006 at The Westin Kuala Lumpur, this years conference promises to once again deliver an International deep-knowledge security conference. HITBSecConf has been described as one of the speakers at last year's conference presented a good paper in SIP (in)security. i believe he'll be returning this year with an advanced paper on VoIP security considerations. p.s. i am one of the organizers for the HITBSecConf series of deep-knowledge security conferences. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR woes
On 03/10/06 05:00 Robert P. McKenzie said the following: Basically the problem is this. While the playbacks are happening you can push any one of the options and to happily goes off and does it. However, if you wait until the messages stop playing back it just hangs up with the error at the bottome of this message. perhaps placing Set(TIMEOUT(response)=XXX) and Set(TIMEOUT(digit)=YYY) at the top of the dialplan would help better. also, bear in mind that these timeouts have to be longer than the time it takes for your IVR voice files to play to be of any good use. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Out IVR
On 03/10/06 19:22 Sharath Chandra said the following: How can i configure the following scenario, - User 'A' dials into Asterisk, - Asterisk puts user 'A' on hold - Dials Out to User 'B' - Consults user B' if he wants to take the call (Press 1) or divert to voicemail (press 2) - Depending on the option chosen, either user A' call is bridged with the out call or transfered to voicemail. have a look at the Privacy() app and the privacy option to the Dial application. they both are able to do what you want. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: MeetMe 'i' option not working correctly?
On 03/09/06 16:41 Tony Mountifield said the following: In article [EMAIL PROTECTED], Jon Webster [EMAIL PROTECTED] wrote: I'm running 2.4.5 and app_meetme never plays conf-hasleft or conf-hasjoined with user names. I looked at app_meetme.c, but couldn't determine the cause. Any suggestions are greatly appreciated. exten = 600,1,MeetMe(600|i) I get the following: -- Executing MeetMe(SIP/jon-21f8, 600|aciMps) in new stack == Parsing '/etc/asterisk/meetme.conf': Found Mar 8 06:13:53 WARNING[5197]: channel.c:2535 ast_request: No channel type registered for 'zap' Mar 8 06:13:53 WARNING[5197]: app_meetme.c:461 build_conf: Unable to open pseudo channel - trying device The above messages indicate that chan_zap.so isn't loaded. Possibly it isn't even built. You need to build *and install* zaptel before starting to build Asterisk. Asterisk will find the zaptel libraries and will build chan_zap. MeetMe requires a timing device, you'd need either a zaptel line card or to load ztdummy to provide pseudo timing. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr data
On 03/09/06 23:04 Dov Bigio said the following:
Hello,
I have an E1 and the possibility to use different caller ids in this E1,
so, before a Dial, I always have a SetCallerIDNum(User, number).
When I check the CDR, the originator of the calls appears to be this
number I set in the caller id, but not the actual user that originated
the call.
the originator of the call, CDR(src) field, is set to the ANI if it's not
null, and to CALLERID(num) if it's null. to get the behaviour you want to
see you'd need to do the following in your dialplan:
Set(CALLERID(ani)=${CALLERID(num)})
Set(CALLERID(num)=)
Dial()
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
| for b in clients employers associates relatives neighbours pets; do |
| echo The opinions here in no way reflect the opinions of my $a $b. |
| done; done |
+=+
___
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[Asterisk-Users] PSTN lines permission settings to different extensions
Note: forwarded message attached. Relax. Yahoo! Mail virus scanning helps detect nasty viruses!---BeginMessage--- Now i will try to explain ..I have 4 PSTNlines in the PBX server 1,2,3,4. Firstline will be usedby only one extension (i.e. for the boss) for incoming and outgoing. This line is dedicated for him only.(No roleovers etc.)The remaining lines will be shared bythe employees 1) Group Ahave access to lines 2 , 3 4. When the employees in Group A dials, then the system checks for the availability of line 4, if available, thenhe can call through that line else (ifthat line is busy) the system checks for the availability of line 3, if available, then he can call through that lineelse (ifthat line is busy) then the system checks for the availability of line 2, if available, then he can call through that line else(ifthat line is busy) then hewill get a busy tone 2)Group Bhave access tolines 2 3 When the employees in GroupB dials, then the system checks for the availability of line 3, if available, thenhe can call through that line else (ifthat line is busy) then the system checks for the availability of line 2, if available, then he can call through that line else(ifthat line is busy) then hewill get a busy tone3)Group C have access to line 2 When the employee in Group C dials then, the system checks for the availability for line 2, if available, then he can call through that line else(ifthat line is busy) then hewill get a busy toneI hope it will be clear now I also want to know how to make these groups. Please explain the answer because i am new in this ... I will be grateful for ur help.Thanks a lot. Faisal Yahoo! Mail Use Photomail to share photos without annoying attachments.---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] gsm picocells
On Sat, 2006-03-18 at 16:49 +1100, James Harper wrote: I believe the OP wants to use GSM handsets as extensions, like running your own localized GSM network. That's not the same as using a GSM terminal to connect Asterisk to the cellular network. Correct! IP Access makes such products. http://www.ipaccess.com/products/nanoBTS.htm That looks about right. All problems of spectrum licensing etc aside, the product claims to use Ethernet as the wired access medium, but appears to need to connect to a much meatier box as part of a packaged solution. The site doesn't seem to give much away, including price. Thanks James Hi James, Was looking also in that direction. Next week (thursday) i will get a delegation from ipaccess and a demo. ( i know it is possible to use wifi/bluetooth phones, but if the intendend target-group can use it's own provate gsm-phone (which they all have), i have a distint feeling those will last longer. ;-) (and cost ME less!)) I will share any info i'll get asap... Hans pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] An FXO version of IAXy?
Steve Murphy wrote: Hello-- In the interest of Symmetry, does anyone else in the world see any need for a device like the IAXy (or the SIP ones from other manufacturers, like the ATA186), but one that presents an FXO interface instead, so it can be connected not to phones, but the PSTN? Absolutely. FXO ata's are the biggest problem I have. I install Astlinux PBX units which work great but we have had to rely on the SPA-3000 for FXO component. An Asterisk specific FXO unit would be excellent. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] An FXO version of IAXy?
In the interest of Symmetry, does anyone else in the world see any need for a device like the IAXy (or the SIP ones from other manufacturers, like the ATA186), but one that presents an FXO interface instead, so it can be connected not to phones, but the PSTN? There's a hugh market for such a box, and none of the current manufacturers have addressed the one-to-four pstn line boxes with anything that would be considered reasonable quality. The GS 488 appears to be their 'test-the-market' box, but its not very usable based on my testing. The Mediatrix 1204 does an excellent job with audio, but is over-priced and under-supported from my perspective. The spa3k comes the closest to providing a reasonable interface with acceptable audio, but has several functions that really need to be fixed. I hope the Linksys folks address those issues instead of dropping the box. From my perspective, designing a fxo box that can interface to the many country standards and has a reasonable echo canceller is not an easy task. Much more difficult than designing a fxs box. And, if you look at the cost of the hardware echo canceller chips that can support 128 taps, the manufacturing cost of a fxo box becomes rather expensive. If you look at the market from a manufacturer's perspective, the sales of fxo boxes are significantly less then the sales of fxs boxes. Therefore it makes sense what the majority of them are doing from an RD and manufacturing perspective (eg, address the larger market before incurring the expense of the smaller market). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] accessing speed dial database
I'm currently running [EMAIL PROTECTED] v 2.7. However I believe asterisk has inbuilt a system wide speed dial system. Preserved number range starting at 300. Just wondering if it's possible to view/backup/restore/modify this data without having to enter it in manually. e.g. 300 301 12345678 (to save phone number 12345678 in speed dial 301?) I'm looking at creating a new installation and need someway of coping over the current database of 100 speed dials. Haven't been able to find any info about this (apart from the documentation explaining how to create speed dials) TIA. Regards, Ian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 - no ring?
Thanks, working great now! Peter Johnson wrote: Have a look in the Polycom phone directory - see if the number of the first 7960 is defined in there with a ring type of 0 (silent). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Sunday, 19 March 2006 1:51 PM To: Asterisk Users-List Subject: [Asterisk-Users] Polycom IP600 - no ring? Have a strange problem... When a C7960 calls the Polycom ip600, the ip600's first line button blinks, the ip600 display shows the proper callerid, but the phone does not ring at all. If I call the same ip600 from a bt102, the ip600 rings properly. If I call the same ip600 from another C7960, the ip600 rings properly. All phones and asterisk are on the same lan within a few feet. The ip600 is running v1.5.2.0054 and works very well in all other respects. Any thoughts on how to diagnose this? (Its definitely not an asterisk config problem.) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk billing
Does anyone know any asterisk billing utilities that would drop the caller back to your own IVR after authentication and still log time used. . . no dial out needed. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] accessing speed dial database
Ian Pilkington wrote: However I believe asterisk has inbuilt a system wide speed dial system. Incorrect. Asterisk has the facilities through the use of the internal (or extenal) database and the dial plan to create such a system, but it's not built-in. Preserved number range starting at 300. You would probably get a quicker answer to this on the AAH mailing list. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
David Rahn wrote: This would lend its self to less repitition of questions as the lists would be much more searchable At this time I 3 months of this list and it is over 13,900 messages. In other words GREAT IDEA I THIRD THAT!! I do think all hardware disscussion ( as it effects Asterisk) should be grouped togeather. As it is not always the exact same problem that is what helps to fix your problem ... Actually, for something like Asterisk, that has so many different aspects, a Forum would be a much better idea. Then, each piece of hardware can have its own category, along with an FAQ. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 and latency measures
Erick Perez wrote: Hi, we have set up a small project in a school the following way: SITE_A(4 port analog to ip g729)--ADSL_ISP1---ISP2Asterisk-PSTN Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit) The asterisk box gets internet service via a wireless antenna. 1 Mbit of up/down bandwith Comments: So far, this means that I will need licenses for the 729. asterisk only supports 20ms sampling on g729 so 4 channels will need 96 kilobits at 20ms sampling (or is it kilobytes??) for the internet bandwith. i cannot use CRTP because i cant be sure if the ISP's routers are CRTP aware. Installing ADSL from ISP1 on the asterisk place will give a clear advantage Please correct any of my prior statements if wrong. should I maintain packet latency below 300ms or 150ms? The objective should be to keep latency as low as possible, however some folks do run asterisk via satellite which as a very lengthy latency. How can I measure this latency all the way to the asterisk? Several ways depending on how accurate a measurement you want. A simple ping would give a starting point. A much more expensive way is to use VoIP analysis software to measure it, but be prepared to spend at least $1,500 (US) to do that. Should I ping from SITE_A to the asterisk box with 8k packets? If you want to emulate a sip/iax packet, use a packet size of about 200 bytes. If I can't install ADSL for the moment, will the above setup work? Probably a bigger issue to address relates to what other traffic might be passing across the dsl and/or wireless channel that might be consuming bandwidth and impacting the rtp packets. Broadcasts originating from devices outside your control (other isp users), hackers attempting to access your ip addresses (at both ends), data traffic between your two endpoints, etc, are just some thoughts of items using a portion of the bandwidth available. Might also think about jitter (eg, variations in latency) and what that might do to your end to end communications. There are other low bandwidth codecs available that could be used instead of g729. Some include ilbc, g726, gsm, etc. Each consumes different bandwidths, and each provide a slightly different quality of audio. See the wiki for more detail on what each consumes for bandwidth on the wire. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Charles Marcus wrote: Actually, for something like Asterisk, that has so many different aspects, a Forum would be a much better idea. Then, each piece of hardware can have its own category, along with an FAQ. You mean something like: http://forums.digium.com/ http://www.voipuser.org/forum_index_5.html http://voxilla.com/forum-viewforum-f-17.html http://www.freeworlddialup.com/community/forum/viewforum.php?f=10 And those are just the ones I frequent on a regular basis. I find the lists are much simpler as I at least get everything in one place. I agree that approx 200-250 messages/day is a high volume, so you have to proactively manage the flow of data. The trick I use is to delete from my mailbox anything that does not seem to be interesting or immediately relevant to my needs. If, at a later date, I need to refer back to the list, I use the archives http://lists.digium.com/mailman/listinfo/ - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBRB2CfktP/KMNOfRbAQL2kwf/XlwAumQEWImkPAuwr1kEwFuJgabEOiV1 t9SsvLdn6gzjwmENM3I54ZvEDIxLbZtYUZTpt8Kh1UT0ukX7ebrBsiFvkG6cXbA8 dLm0Shar048pgmbufT8gvws6gpJaSijAniVPhmJV3qEzzjkk6wmZfab3KOavNWMH y1hzOz1dQyr0qmagdStzKvvwaPCKNuRMiItM8lb+uVAgB9z0BkJjSCT7xpXXuz9Y axlaHA2k07WG4vgsUytPvoLqRc4R09Wt34Kznt5MvXWSw4puXjHkl3DqWSnPuC3Q Wadp9oOiUrpaia6P3Gq1HvvwGSJ8LW6o09FArhCi2VLYE9ztz6Gv6Q== =gzVa -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sending ANI to TDM40B FXS?
We are using TDM40B's to connect some devices to Asterisk which depend on caller information arriving as ANI, rather than as Caller ID. I am unsure if the TDM40B supports this in the first place, and if so, I am unsure how to configure it so. I've searched the wiki but couldn't find anything. Can someone please confirm whether or not this is possible? As a fallback I could reconfigure my FXS analog device to use Caller ID but I would rather leave it as ANI if possible. Thank you, Bryan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 and latency measures
Thanks Rich, but i'm only allowed to use g729. you said that some folks run high latency connections, but is 300ms high in my setup? On 3/19/06, Rich Adamson [EMAIL PROTECTED] wrote: Erick Perez wrote: Hi, we have set up a small project in a school the following way: SITE_A(4 port analog to ip g729)--ADSL_ISP1---ISP2Asterisk-PSTN Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit) The asterisk box gets internet service via a wireless antenna. 1 Mbit of up/down bandwith Comments: So far, this means that I will need licenses for the 729. asterisk only supports 20ms sampling on g729 so 4 channels will need 96 kilobits at 20ms sampling (or is it kilobytes??) for the internet bandwith. i cannot use CRTP because i cant be sure if the ISP's routers are CRTP aware. Installing ADSL from ISP1 on the asterisk place will give a clear advantage Please correct any of my prior statements if wrong. should I maintain packet latency below 300ms or 150ms? The objective should be to keep latency as low as possible, however some folks do run asterisk via satellite which as a very lengthy latency. How can I measure this latency all the way to the asterisk? Several ways depending on how accurate a measurement you want. A simple ping would give a starting point. A much more expensive way is to use VoIP analysis software to measure it, but be prepared to spend at least $1,500 (US) to do that. Should I ping from SITE_A to the asterisk box with 8k packets? If you want to emulate a sip/iax packet, use a packet size of about 200 bytes. If I can't install ADSL for the moment, will the above setup work? Probably a bigger issue to address relates to what other traffic might be passing across the dsl and/or wireless channel that might be consuming bandwidth and impacting the rtp packets. Broadcasts originating from devices outside your control (other isp users), hackers attempting to access your ip addresses (at both ends), data traffic between your two endpoints, etc, are just some thoughts of items using a portion of the bandwidth available. Might also think about jitter (eg, variations in latency) and what that might do to your end to end communications. There are other low bandwidth codecs available that could be used instead of g729. Some include ilbc, g726, gsm, etc. Each consumes different bandwidths, and each provide a slightly different quality of audio. See the wiki for more detail on what each consumes for bandwidth on the wire. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 and latency measures
Yes, 300ms seems very high if there is no satellite link involved. g729 should be just fine if that's what you're stuck with. Erick Perez wrote: Thanks Rich, but i'm only allowed to use g729. you said that some folks run high latency connections, but is 300ms high in my setup? On 3/19/06, Rich Adamson [EMAIL PROTECTED] wrote: Erick Perez wrote: Hi, we have set up a small project in a school the following way: SITE_A(4 port analog to ip g729)--ADSL_ISP1---ISP2Asterisk-PSTN Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit) The asterisk box gets internet service via a wireless antenna. 1 Mbit of up/down bandwith Comments: So far, this means that I will need licenses for the 729. asterisk only supports 20ms sampling on g729 so 4 channels will need 96 kilobits at 20ms sampling (or is it kilobytes??) for the internet bandwith. i cannot use CRTP because i cant be sure if the ISP's routers are CRTP aware. Installing ADSL from ISP1 on the asterisk place will give a clear advantage Please correct any of my prior statements if wrong. should I maintain packet latency below 300ms or 150ms? The objective should be to keep latency as low as possible, however some folks do run asterisk via satellite which as a very lengthy latency. How can I measure this latency all the way to the asterisk? Several ways depending on how accurate a measurement you want. A simple ping would give a starting point. A much more expensive way is to use VoIP analysis software to measure it, but be prepared to spend at least $1,500 (US) to do that. Should I ping from SITE_A to the asterisk box with 8k packets? If you want to emulate a sip/iax packet, use a packet size of about 200 bytes. If I can't install ADSL for the moment, will the above setup work? Probably a bigger issue to address relates to what other traffic might be passing across the dsl and/or wireless channel that might be consuming bandwidth and impacting the rtp packets. Broadcasts originating from devices outside your control (other isp users), hackers attempting to access your ip addresses (at both ends), data traffic between your two endpoints, etc, are just some thoughts of items using a portion of the bandwidth available. Might also think about jitter (eg, variations in latency) and what that might do to your end to end communications. There are other low bandwidth codecs available that could be used instead of g729. Some include ilbc, g726, gsm, etc. Each consumes different bandwidths, and each provide a slightly different quality of audio. See the wiki for more detail on what each consumes for bandwidth on the wire. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Annoying Asterisk Realtime Limitation
Well, this is a major pain in the ass. I got realtime static working for sip.conf. 'Great!' I thought. That was until I realised I couldn't use it. Our Asterisk systems are using OSPF and listen on interface lo:1. Asterisk doesn't like to use an interface name for it's bindaddr setting, so you have to put the IP address of lo:1 in there. If you put in 0.0.0.0, it seems to listen on the first interface it finds, probably eth0. You can't do that with OSPF because it's load balancing and traffic can come over eth1 instead. If you point all your Asterisk systems to a single table for sip.conf, what do you put in the binaddr setting? You can't use the systems IP address at lo:1, because they're all different, and you can't use 0.0.0.0 or lo:1. The only solution is to have one sip.conf table for every Asterisk system... 5 in our case. Anyway, so I went back to a plain text file for sip.conf. What a dissapointment. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bizzare DTMF on channel bank
I have incoming PSTN lines on an Adtran 750 channel bank. Calls are
evaluated by an agi script based on callerid and forwarded to an
international DID through Voxee. There is an IVR at that number that
asked to user to enter a selection. When the user presses a key, my pbx
puts the call on hold and tries to start music on hold. What's doing
this? I have no backgrounds, no listen, the call should be completed and
asterisk should not be listening by then.
-- Executing Dial(Zap/1-1, IAX2/voxee/1901234|60|tr) in new
stack
-- Called voxee/1901234
-- Call accepted by 66.246.246.52 (format g729)
-- Format for call is g729
-- IAX2/voxee-16384 is ringing
-- IAX2/voxee-16384 answered Zap/1-1
-- Started music on hold, class 'default', on channel 'Zap/1-1'
-- Playing 'pbx-transfer' (language 'en')
-- Stopped music on hold on Zap/1-1
-- Unable to find extension '' in context 'default'
-- Playing 'pbx-invalid' (language 'en')
[pstn-in]
;
exten = s,1,Answer()
exten = s,n,NoOp(Acount: ${ACCOUNTCODE} - Ext: ${EXTEN} - CallerID:
${CALLERID})
exten = s,n,agi,call.agi
exten = s,n,NoOp(Call from ${CALLERID} to ${DID} Account ${ACCOUNT})
exten = s,n,Dial(IAX2/voxee/${DID},60,tr)
exten = s,n,Hangup
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
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dangerous content by MailScanner, and is
believed to be clean.
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[Asterisk-Users] Voicemail Bug?
Ugh. I have voicemail set up for realtime... mysql SELECT * FROM ast_vm_users; +--+-+---+-+--+--+---+---+-+ | uniqueid | customer_id | context | mailbox | password | fullname | email | pager | stamp | +--+-+---+-+--+--+---+---+-+ |1 | 0 | voicemail | 3254104 | 1234 | | | | 2006-03-13 13:35:12 | |2 | 0 | voicemail | 3254101 | 1234 | | | | 2006-03-13 13:42:46 | |3 | 0 | voicemail | 3254102 | 1234 | | | | 2006-03-13 13:42:46 | |4 | 0 | voicemail | 2944093 | 1234 | | | | 2006-03-13 13:42:46 | +--+-+---+-+--+--+---+---+-+ I have extconfig set up for realtime voicemail: voicemail = mysql,vox180internal,ast_vm_users I enter the voicemail system through VoiceMailMain(). I enter my password. I then enter option 3 for advanced options, and then option 5 to leave a message (I'm assuming this lets me leave a message for another mailbox). Asterisk asks me for the mailbox number. I enter 3254104. The following appears on the console: Mar 19 11:34:33 WARNING[30855]: app_voicemail.c:2411 leave_voicemail: No entry in voicemail config file for '3254104' I run ngrep on the database and watch the queries. Asterisk sends this... SELECT * FROM ast_vm_users WHERE mailbox = '3254104' AND context = 'voicemail' So what's it's problem? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 and latency measures
Erick Perez wrote: How can I measure this latency all the way to the asterisk? I have found two good ways to monitor routes for VOIP. Install mtr and run mtr your.voipserver to find where you are seeing the latency, and then install smokeping (not so easy to install) and you will be able to monitor the latency over time. I find smokeping the most reliable way to visually gauge route quality. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel will not build
FYI: I am trying to build zaptel-1.2.4 against the recently updated kernel version 2.6.9-34.EL on Centos 4.2. but I am getting errors and it will not build. This is apparently due to a typo in a kernel header spinlock.h although I have not successfully modified the kernel and built zaptel against it yet. https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568 This bug report has a typo as well. It should read: #define DEFINE_RWLOCK(x) rwlock_t x = RW__LOCK_UNLOCKED make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /usr/src/zaptel-1.2.4/zaptel.o /usr/src/zaptel-1.2.4/zaptel.c:384: error: syntax error before zone_lock /usr/src/zaptel-1.2.4/zaptel.c:384: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel-1.2.4/zaptel.c:384: error: incompatible types in initialization /usr/src/zaptel-1.2.4/zaptel.c:384: error: initializer element is not constant /usr/src/zaptel-1.2.4/zaptel.c:384: warning: data definition has no type or storage class /usr/src/zaptel-1.2.4/zaptel.c:385: error: syntax error before chan_lock /usr/src/zaptel-1.2.4/zaptel.c:385: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel-1.2.4/zaptel.c:385: error: incompatible types in initialization /usr/src/zaptel-1.2.4/zaptel.c:385: error: initializer element is not constant /usr/src/zaptel-1.2.4/zaptel.c:385: warning: data definition has no type or storage class /usr/src/zaptel-1.2.4/zaptel.c:188: warning: 'fcstab' defined but not used make[2]: *** [/usr/src/zaptel-1.2.4/zaptel.o] Error 1 make[1]: *** [_module_/usr/src/zaptel-1.2.4] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-i686' make: *** [linux26] Error 2 -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
At 07:50 AM 03/19/2006, you wrote: Actually, for something like Asterisk, that has so many different aspects, a Forum would be a much better idea. Then, each piece of hardware can have its own category, along with an FAQ. There's lots of Asterisk forums out there already, but weirdly enough all the really good information is in the mailing list. Funny how that seems to be a consistent pattern in the world. Ira -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.385 / Virus Database: 268.2.5/284 - Release Date: 03/17/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
On 19 Mar 2006, at 17:47, Ira wrote: At 07:50 AM 03/19/2006, you wrote: Actually, for something like Asterisk, that has so many different aspects, a Forum would be a much better idea. Then, each piece of hardware can have its own category, along with an FAQ. There's lots of Asterisk forums out there already, but weirdly enough all the really good information is in the mailing list. Funny how that seems to be a consistent pattern in the world. Whether or not a forum is a better idea isn't really depending on the subject matter IMHO. Its success or failure depends on what the prospective participants like better. I personally cannot stand forums. That's a place where I have to expend energy to go there and manually click through stuff. If I remember to go there and say up to date, that is. Email comes to me, and is sorted suitably on the server side so there is no clutter. Deleting messages I don't care about is much easier than clicking myself through some thread on a forum. jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Annoying Asterisk Realtime Limitation
On 19 Mar 2006, at 17:56, Douglas Garstang wrote: Well, this is a major pain in the ass. I got realtime static working for sip.conf. 'Great!' I thought. That was until I realised I couldn't use it. Our Asterisk systems are using OSPF and listen on interface lo:1. Asterisk doesn't like to use an interface name for it's bindaddr setting, so you have to put the IP address of lo:1 in there. If you put in 0.0.0.0, it seems to listen on the first interface it finds, probably eth0. You can't do that with OSPF because it's load balancing and traffic can come over eth1 instead. If you point all your Asterisk systems to a single table for sip.conf, what do you put in the binaddr setting? You can't use the systems IP address at lo:1, because they're all different, and you can't use 0.0.0.0 or lo:1. The only solution is to have one sip.conf table for every Asterisk system... 5 in our case. At the risk of stirring up a flame war. If you have a 'real' database you could work around that problem with a view. The view could be written to pass back a different value of bindaddr depending on which client asks, but all the other values come straight out of base table that is the same for all clients. A bind addr of 0.0.0.0 should listen on all interfaces that are up at the time the listen is started, I guess your problem is with the source address in the outgoing (from asterisk) reply packets. Anyway, so I went back to a plain text file for sip.conf. What a dissapointment. I do think there might have been a work around available there. Doug. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] An FXO version of IAXy?
YES YES YES -Original Message- From: Steve Murphy [mailto:[EMAIL PROTECTED] Sent: Sunday, March 19, 2006 2:32 AM To: Asterisk User List Subject: [Asterisk-Users] An FXO version of IAXy? Hello-- In the interest of Symmetry, does anyone else in the world see any need for a device like the IAXy (or the SIP ones from other manufacturers, like the ATA186), but one that presents an FXO interface instead, so it can be connected not to phones, but the PSTN? murf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calls to SIP providers
Hi All, Which configuration settings will allow my Asterisk registered softphones to make calls to third party SIP provider subscribers (fwd, voiptalk, etc.) by directly entering the sip uri (e.g. sip:[EMAIL PROTECTED])? I have googled for configurations, but they all assume that Asterisk is registered with a provider and the provider routes the calls with dial prefixes. I want to bypass any provider and go straight to the external phone sip address. In my current configuration I get an immediate 'Number does not exist Call rejected: 404 Not Found' when I try to call an external sip phone. Judging by the speed of the error, I assume that asterisk looks for internal users only. Do I need to add a line in extension.conf that tells asterisk where to route SIP calls? I have seen lines like these: exten = _sip.1, If so, what action should I put on such a line? Bart... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] trunking questions
Is there a non hardware limit to the limit of concurrent connections that can go over a trunk? So IAX trunking is preferred, can * do any other trunking? Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem w/ Dial Command on Zap channel
I'm running Asterisk 1.2 on RedHat 9.0 and found he following problem with the Dial command when used on a Zap channel:
When I have this in my dial plan everything works fine:
exten = _9NXX,1,Dial(ZAP/4/${EXTEN:1})
However I'd like to include the 't' option, but when I do either of the following:
exten = _9NXX,1,Dial(ZAP/4/${EXTEN:1},,t)
or
exten = _9NXX,1,Dial(ZAP/4/${EXTEN:1},30,t)
it doesn't work. The output from CLI is shown below:
-- Called 4/7451576
-- Zap/4-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/4-1
It looks OK, but I cannot hear anything. Only the firs Dial command seems to work.
Anyone else run into this? Is there a work around?
Thanks,
Hugh
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[Asterisk-Users] Grandstream unit HT-488
Hi All, Anybody knows how to terminated calls using Grandstream Ht488 and the FXO port ? I can ring the FXO port fine , rings 1once then give me dial tone. Thanks, Oliver Vermeulen World Venture Group Telecom Tech / Admin Corporate Address: Str Avionului Nr 35/bl16J/3 Bucharest, 014333 Romania Tel Romania: +(40) 31-860-0030 Fax: +(40) 31-860-0031 USA DID: + 1 (305) 722-1457 DR DID: +1(809) 202-6932 BELGIUM DID: +(32) 9 395-5620 UK DID: +(44) 870-478-8896 SIP : [EMAIL PROTECTED] website : http://www.wvg-tele.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN NT Mode CAPI
I'm setting up an asterisk server to allow our PBX to make calls out via VoIP, but when it calls out I get this message: chan_capi.c: did not find device for msn = (eg no msn) Which would be correct because at that point I've only asked for an outside line. I'm using CAPI obviously, and my config is: [ISDN1] ntmode=yes isdnmode=did incomingmsn=* immediate=yes controller=1 group=1 softdtmf=on relaxdtmf=on accountcode= context=default devices=2 Any pointers? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup Woes
Hello all, I have an asterisk @ home system running 1.2.4. Call pickup seems to be a bit of a problem. Ive looked at a lot of posts and the wiki, which states that you need to define the pickup extension in features.conf and the pickup groups in sip.conf. Ive done this, however there is no definition for *8 in extensions.conf. Is there supposed to be and it has been removed? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
There shouldn't be one, have you tried it? what is the CLI output? On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: Hello all, I have an asterisk @ home system running 1.2.4. Call pickup seems to be a bit of a problem. I've looked at a lot of posts and the wiki, which states that you need to define the pickup extension in features.conf and the pickup groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Is there supposed to be and it has been removed? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Users Mailing List Traffic
Actually, for something like Asterisk, that has so many different aspects, a Forum would be a much better idea. Then, each piece of hardware can have its own category, along with an FAQ. Please no. A forum might be okay if you have a nice fast web connection and/or a bit of patience, and if you only subscribe to the one list. I subscribe to quite a few lists these days and the mere thought of having to go and visit a web site for each one to read them almost brings tears to my eyes :) That being said, a mailing list with a forum interface (or a forum with a mailing list option) might be a reasonable compromise as it should meet the needs of both mailing list lovers and forum lovers (assuming it is implemented properly!) James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer to specific park number
HiI'd like to allow users to transfer a call to a specific park number. This way, the receptionist can tranfer a call park for ext 100 at park number 7100 etc...It seems like this should be fairly simple using the Park(ext) app but it doesn't work for me. No matter what I extension I use, the system just picks the next available park number. I've simplified my dialplan for testing. Here's what I'm working with now
Features.conf--[general]parkext = 70 parkpos = 7000-7010context = parkedcallscourtesytone = beep xfersound = beep xferfailsound = beeperr
findslot = first
pickupexten = *8featuredigittimeout = 500[featuremap]blindxfer = #1 automon = *1atxfer = *2 Extensions.conf[general]
static=yeswriteprotect=yesautofallthrough=yesclearglobalvars=nopriorityjumping=no[globals]CONSOLE=Console/dsp ; Console interface for demoIAXINFO=guest ; IAXtel username/password
[macro-stdexten];exten = s,1,Dial(${ARG2},10,rtwTW)exten = s,2,Goto(s-${DIALSTATUS},1)exten = s-NOANSWER,1,Voicemail(u${ARG1})exten = s-NOANSWER,2,Goto(default,s,1)exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)exten = _s-.,1,Goto(s-NOANSWER,1)exten = a,1,VoicemailMain(${ARG1})[internal]include = parkedcallsexten = 100,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = 101,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})exten = 102,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})exten = 103,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})exten = 104,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = 123,1,Answer()exten = 123,2,Park(7002)If I transfer a call to 123, it parks it on 7000. Here's the output from the console. -- Executing Macro(SIP/100-fd8c, stdexten|101|SIP/101) in new stack
-- Executing Dial(SIP/100-fd8c, SIP/101|10|rtwTW) in new stack -- Called 101 -- SIP/101-85d6 is ringing -- SIP/101-85d6 answered SIP/100-fd8c -- Attempting native bridge of SIP/100-fd8c and SIP/101-85d6
Asterisk1*CLI Asterisk1*CLI -- Started music on hold, class 'default', on channel 'SIP/100-fd8c' -- Stopped music on hold on SIP/100-fd8c -- Executing Answer(SIP/101-16b7, ) in new stack
-- Executing Park(SIP/101-16b7, 7002) in new stack == Parked SIP/101-16b7 on 7000. Will timeout back to extension [internal] s, 1 in 45 seconds -- Added extension '7000' priority 1 to parkedcalls
-- Playing 'digits/7' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/0' (language 'en') -- Started music on hold, class 'default', on channel 'SIP/101-16b7'
== Spawn extension (internal, s, 1) exited KEEPALIVE on 'SIP/101-16b7' I'm using Asterisk 1.2.5. I'm fairly new to asterisk so it's possible I'm missing something simple. Any suggestions would be appreciated.
Thanks-- kris seraphine
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RE: [Asterisk-Users] Annoying Asterisk Realtime Limitation
No flames here as I realize that there are plenty of limitations with MySQL, but if you're using the current GA of it views is not one of them. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Sunday, March 19, 2006 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Annoying Asterisk Realtime Limitation On 19 Mar 2006, at 17:56, Douglas Garstang wrote: Well, this is a major pain in the ass. I got realtime static working for sip.conf. 'Great!' I thought. That was until I realised I couldn't use it. Our Asterisk systems are using OSPF and listen on interface lo:1. Asterisk doesn't like to use an interface name for it's bindaddr setting, so you have to put the IP address of lo:1 in there. If you put in 0.0.0.0, it seems to listen on the first interface it finds, probably eth0. You can't do that with OSPF because it's load balancing and traffic can come over eth1 instead. If you point all your Asterisk systems to a single table for sip.conf, what do you put in the binaddr setting? You can't use the systems IP address at lo:1, because they're all different, and you can't use 0.0.0.0 or lo:1. The only solution is to have one sip.conf table for every Asterisk system... 5 in our case. At the risk of stirring up a flame war. If you have a 'real' database you could work around that problem with a view. The view could be written to pass back a different value of bindaddr depending on which client asks, but all the other values come straight out of base table that is the same for all clients. A bind addr of 0.0.0.0 should listen on all interfaces that are up at the time the listen is started, I guess your problem is with the source address in the outgoing (from asterisk) reply packets. Anyway, so I went back to a plain text file for sip.conf. What a dissapointment. I do think there might have been a work around available there. Doug. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
You have to configre the Dialplan in your sip phone to accept *8 What phone are you using? On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HFC USB (was MultiBRI in Australia - found one - maybe)
Hmm, I was using 0.3.0 rc24, or the unstable branch. I see 0.2.0 is listed as 'stable' so maybe I should have used that. Please do keep me informed of your progress. Craig After finally getting chan_misdn to load (missing #include to bitops.h under Debian at least) it still won't load, and won't tell me why even with all the debug stuff turned on. 0.3.0rc25 is what I'm using. chan_capi works in TE mode, but I can't get it working in NT mode which is what I want (keeps complaining about not being able to find a device for a blank msn). Could you please post something about what you did to get chan_misdn going? I have an idea that I've got a bad version of something compiled somewhere but hopefully it is solvable. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
I am using Cisco 7940/60/70's -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes You have to configre the Dialplan in your sip phone to accept *8 What phone are you using? On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes You have to configre the Dialplan in your sip phone to accept *8 What phone are you using? On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
C F wrote: Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's Don't you mean the dialplan.xml. This is what I have: DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- TEMPLATE MATCH=#Timeout=5/ !-- Anything else -- /DIALTEMPLATE -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
Thank you very much. I'll now investigate how to set up dialplan.xml. I've never had to set it up before. Cheers, Much appreciated. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes You have to configre the Dialplan in your sip phone to accept *8 What phone are you using? On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HFC USB (was MultiBRI in Australia - found one -maybe)
Hmm, I was using 0.3.0 rc24, or the unstable branch. I see 0.2.0 is listed as 'stable' so maybe I should have used that. Please do keep me informed of your progress. Craig After finally getting chan_misdn to load (missing #include to bitops.h under Debian at least) it still won't load, and won't tell me why even with all the debug stuff turned on. 0.3.0rc25 is what I'm using. chan_capi works in TE mode, but I can't get it working in NT mode which is what I want (keeps complaining about not being able to find a device for a blank msn). Could you please post something about what you did to get chan_misdn going? I have an idea that I've got a bad version of something compiled somewhere but hopefully it is solvable. James - OK Being the OH so Lazy person that I am...here are the steps that I took to get this all going. Started with my Stock Standard CentOS 4.2 install ... Installed 2.6.11 Kernel sources. Compiled and installed as per normal...turning off spinlock_debug and SMP Rebooted into new kernel. Installed mISDN using the install_misdn script Recompiled zaptel (for the hell of it...and so that I had a timming source) Manually setup the /etc/misdn-init.conf The autodiscovery thing didn't pickup the devices. Added the following three lines to my rc.local rmmod hfc_usb rmmod hisax /etc/init.d/misdn-init start Reboot once moreand that was it Dave -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.2.5/284 - Release Date: 17/03/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
in AAH you can set the callgroup and pickup group within each extensions setup. On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: Thank you very much. I'll now investigate how to set up dialplan.xml. I've never had to set it up before. Cheers, Much appreciated. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes You have to configre the Dialplan in your sip phone to accept *8 What phone are you using? On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to receive fax
Hi Thanks for your reply and recommendation. Can you tell me more detail of how to do it as I am really fresh in Asterisk. Again, thank you. Gidean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
H, I'm still a little stumped. I edited SIPDefault to and created a dialplan.xml file which is being uploaded to the phone. Still no output on the asterisk console wheh I dial *8. :( dialplan.xml DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- /DIALTEMPLATE SIPDefault.cnf extract: # XML file that specifies the dialplan desired dial_template: dialplan :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, 20 March 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's Don't you mean the dialplan.xml. This is what I have: DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- TEMPLATE MATCH=#Timeout=5/ !-- Anything else -- /DIALTEMPLATE -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GS BT102 dual ethernet port -bandwidth impact
The bt102 is a 10megabit switch so I don't get what you are saying? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Saturday, March 18, 2006 8:43 PM To: Asterisk Users-List Subject: [Asterisk-Users] GS BT102 dual ethernet port -bandwidth impact FYI for anyone using the dual ethernet ports on a Grandstream BT102. I'm using a BT102 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC - HP Switch - Asterisk : actual throughput measured at 94.1 mbps. PC - BT102 - HP Switch - Asterisk : actual measured at 8.86 mbps. The second test (through the BT102) was conducted with a g711 conversation in progress. Audio quality was noticeably impacted presumably due to the half duplex support in the BT102. The BT102 was running sip v1.0.5.18 firmware. The bandwidth tester (older version of NetIQ's QCheck) sent one megabyte bursts of tcp traffic between the two endpoints using 1514 bytes packets. The tests were run purely to document throughput of the phone when used with an attached PC. Rich ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
You don't need to mess with the dialplan.xml on a cisco phone. Try dialing *8# to pick up a ringing phone. It works just fine here with nothing special in features.conf or extensions.conf. Adam Dale wrote: H, I'm still a little stumped. I edited SIPDefault to and created a dialplan.xml file which is being uploaded to the phone. Still no output on the asterisk console wheh I dial *8. :( dialplan.xml DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- /DIALTEMPLATE SIPDefault.cnf extract: # XML file that specifies the dialplan desired dial_template: dialplan :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, 20 March 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's Don't you mean the dialplan.xml. This is what I have: DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- TEMPLATE MATCH=#Timeout=5/ !-- Anything else -- /DIALTEMPLATE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GS BT102 dual ethernet port -bandwidth impact
Its a 10 meg half-duplex switch in the phone. When the ethernet utilization approaches 80% (or more), packets are dropped, impacting the voice conversation. What that implies is not connecting a PC to the second jack if that PC bursts any significant amount of data traffic. Nothing more, nothing less. Michael J. Liberatore wrote: The bt102 is a 10megabit switch so I don't get what you are saying? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Saturday, March 18, 2006 8:43 PM To: Asterisk Users-List Subject: [Asterisk-Users] GS BT102 dual ethernet port -bandwidth impact FYI for anyone using the dual ethernet ports on a Grandstream BT102. I'm using a BT102 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC - HP Switch - Asterisk : actual throughput measured at 94.1 mbps. PC - BT102 - HP Switch - Asterisk : actual measured at 8.86 mbps. The second test (through the BT102) was conducted with a g711 conversation in progress. Audio quality was noticeably impacted presumably due to the half duplex support in the BT102. The BT102 was running sip v1.0.5.18 firmware. The bandwidth tester (older version of NetIQ's QCheck) sent one megabyte bursts of tcp traffic between the two endpoints using 1514 bytes packets. The tests were run purely to document throughput of the phone when used with an attached PC. Rich ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
Unfortunatly I get a beeping sound and that's it. Just like when I dial something that does not have a match in extensions.conf :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, 20 March 2006 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes You don't need to mess with the dialplan.xml on a cisco phone. Try dialing *8# to pick up a ringing phone. It works just fine here with nothing special in features.conf or extensions.conf. Adam Dale wrote: H, I'm still a little stumped. I edited SIPDefault to and created a dialplan.xml file which is being uploaded to the phone. Still no output on the asterisk console wheh I dial *8. :( dialplan.xml DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- /DIALTEMPLATE SIPDefault.cnf extract: # XML file that specifies the dialplan desired dial_template: dialplan :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, 20 March 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's Don't you mean the dialplan.xml. This is what I have: DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- TEMPLATE MATCH=#Timeout=5/ !-- Anything else -- /DIALTEMPLATE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Xorcom TS-1 T1 installs?
Has anyone done any installations using the Xorcom TS-1 and multiple T1's? I'm looking for a reliable box to put in a closet and route calls between T1's. No voicemail, IVR, etc. Any other suggestions? Mark Willis Cartama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk RealTime Question, Please help
Hi Benchev,
Thanks a lot for your replies.
I understood that without mentioning context names in Extensions.conf
we cannot
configure contexts in Asterisk Realtime.
Thanks and Regards,
Manoj.
Quoting Benchev [EMAIL PROTECTED]:
I need many contexts because I have around 1000 DID's each with 5-10
Extensions.
These DID numbers are changed or added very frequently and whenever
there is a change I have to change Extensions.conf manually. So please
tell me how can I do this dynamically without changing Extensions.conf
and help me configure Asterisk.
I presume you have about 1000 DID numbers and each of this numbers
may ring to
5-10 users of yours, right?
If so, make a context in you extensions.conf and include in it a switch
like that:
[ever_changing_dids]
switch = Realtime/[EMAIL PROTECTED]
Now you can insert in your extensions_table imaginary DID 9876543210:
INSERT INTO `extensions_table` VALUES ('', 'ever_changing_dids',
'9876543210',
1, 'Dial', 'SIP/user1:SIP/user2:SIP/user3:SIP/user4:SIP/user8:SIP/user12|
20');
You can do that for many thousands of DIDs without changing
extensions.conf.
My current setup is exactly similar to which you have suggested. My DID
numbers
are added or changed very frequently and all the time I have to change some
config file manually and should reload Asterisk or atleast call Extensions
reload. I do want these things to be manual, Can't I have the Asterisk to
directly get the contexts from Mysql DB without giving them in config
files? If
this is possible then we can have a realtime dynamic Asterisk.
The other approcah can be to match the context itself with some regular
expression. But I do not know how do this or whether this is possible? I
will have a context something like this
[XX]
switch = Realtime/@extensions
So all contexts will be directed to Mysql DB matching regex but
[XX] is
not acting as regex as expected it just matches context XX.
I am afraid I'm loosing you.
However, try to change in extconfig.conf
extentions = mysql,astcc,extensions_table
to:
exten_sions = mysql,astcc,extensions_table
and then switch = Realtime/@exten_sions
Then in extensions.conf:
[inbound]
switch = Realtime/@exten_sions
In mysql do:
INSERT INTO `extensions_table` VALUES ('', 'inbound',
'9876543210', 1, 'Dial', 'SIP/user1|20');
*CLI show dialplan inbound
should show: Alt. Switch = 'Realtime/@exten_sions'
not what you have used to see with a static extensions.conf but you can do:
server*CLI realtime load exten_sions context inbound
Column Name Column Value
id 1
context inbound
exten 9876543210
priority 1
app Dial
appdata SIP/user1|20
and see how realtime took that.
On the other hand, no you can not create contexts on the fly and
out of nothing. Actually the context scheme is the backbone of your system
and should be thought over and set up beforehand.
The connection between particular context and realtime is the switch.
When you insert into the extensions_table a set, which context corresponds
to where the switchis, and this is read in realtime without the need of
reloading.
Pretty much that's all I can do to help. Sorry.
Benchev
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[Asterisk-Users] Local Channel
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I'm using the Local channel in an app of mine and I'm finding that the app is being cut out of the call path. You used to be able to avoid this using the \n command but that doesn't seem to work any more. This is on a recent version of Asterisk. Any comments/suggestion? Darren Wiebe [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEHjLG4DADnh+tnOQRArn+AJ0dx9fncjX77QVtP0VzCXqa2i0BXwCdFv1v 0UQ9s6cloDFZJwIiBWJe/Hg= =fi8U -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feedback from VON expo! Info on * HAandPolycomphone!!
Huh? Phones do a NAPTR/SRV lookup in a specified domain to get a list of SRV records to use. The phones don't query the DNS server every time they make a call... they have a cache. You also run primary and a secondary (or two primary) dns servers. It's a simple scalable solution. It's a shame Asterisk doesn't support it. The only thing Asterisk would use an SRV lookup for is handing calls off to a carrier for termination right? The phones are the ones that need the SRV to try to have a redundant setup and I believe there is nothing wrong at all on relying on this - the whole world relies on DNS and since its inception, there has never been a total failure of the DNS networkpretty reliable if you ask me. For my Polycom, this is a great setup. The only thing is I want to be sure I understand the statement above because the only time I can see Asterisk needing to do an SRV lookup is if it is handing a call to a carrier for termination. - Gabe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 and latency measures
Hi what is mtr ? where can i find that ram On 3/20/06, Chris Mason (Lists) [EMAIL PROTECTED] wrote: Erick Perez wrote: How can I measure this latency all the way to the asterisk? I have found two good ways to monitor routes for VOIP. Install mtr andrun mtr your.voipserver to find where you are seeing the latency, andthen install smokeping (not so easy to install) and you will be able to monitor the latency over time. I find smokeping the most reliable way tovisually gauge route quality.--Chris MasonNetConcepts(264) 497-5670 Fax: (264) 497-8463Int:(305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271Cell: 264-235-5670Yahoo IM: [EMAIL PROTECTED]--This message has been scanned for viruses anddangerous content by MailScanner, and is believed to be clean.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local Channel
Which specific version of Asterisk is this? What was the last version of Asterisk you used that this worked for you? MATT--- On 3/19/06, Darren Wiebe [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I'm using the Local channel in an app of mine and I'm finding that the app is being cut out of the call path. You used to be able to avoid this using the \n command but that doesn't seem to work any more. This is on a recent version of Asterisk. Any comments/suggestion? Darren Wiebe [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEHjLG4DADnh+tnOQRArn+AJ0dx9fncjX77QVtP0VzCXqa2i0BXwCdFv1v 0UQ9s6cloDFZJwIiBWJe/Hg= =fi8U -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HFC USB (was MultiBRI in Australia - found one-maybe)
I've just found my first problem. /dev/mISDN was being created with the wrong permissions... Thanks James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of David Phelan Sent: Monday, 20 March 2006 12:18 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] HFC USB (was MultiBRI in Australia - found one-maybe) Hmm, I was using 0.3.0 rc24, or the unstable branch. I see 0.2.0 is listed as 'stable' so maybe I should have used that. Please do keep me informed of your progress. Craig After finally getting chan_misdn to load (missing #include to bitops.h under Debian at least) it still won't load, and won't tell me why even with all the debug stuff turned on. 0.3.0rc25 is what I'm using. chan_capi works in TE mode, but I can't get it working in NT mode which is what I want (keeps complaining about not being able to find a device for a blank msn). Could you please post something about what you did to get chan_misdn going? I have an idea that I've got a bad version of something compiled somewhere but hopefully it is solvable. James - OK Being the OH so Lazy person that I am...here are the steps that I took to get this all going. Started with my Stock Standard CentOS 4.2 install ... Installed 2.6.11 Kernel sources. Compiled and installed as per normal...turning off spinlock_debug and SMP Rebooted into new kernel. Installed mISDN using the install_misdn script Recompiled zaptel (for the hell of it...and so that I had a timming source) Manually setup the /etc/misdn-init.conf The autodiscovery thing didn't pickup the devices. Added the following three lines to my rc.local rmmod hfc_usb rmmod hisax /etc/init.d/misdn-init start Reboot once moreand that was it Dave -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.2.5/284 - Release Date: 17/03/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Attended Transfer - transfer timeout, how to change?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... you are using the attended transfer feature.. ist it already possible to hang up before the other person lifts the handset without loosing the caller when you are doing an attendet transfer? (person A takes an incoming call, person A would like to do an attended transfer to person B, person A hangs up the phone BEFORE person B takes the transfered call -- does the incoming call get lost?) this was an issue in 1.2.4, I'd like to know whether its fixed in 1.2.5. You shouldn't hang up. You should use disconnect = #0 from features.conf -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream unit HT-488
On Mar 19, 2006, at 2:04 PM, Oliver Vermeulen wrote: x-tad-smallerHi All,/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerAnybody knows how to terminated calls using Grandstream Ht488 and the FXO port ?/x-tad-smallerx-tad-smallerI can ring the FXO port fine , rings 1once then give me dial tone./x-tad-smallerx-tad-smaller /x-tad-smaller I had: exten => _NXX,1,Dial(SIP/@2003,60,D(w$EXTEN})) exten => _NXX,2,Hangup Where 2003 was the extension of the FXO on the HT-488. This worked ok for dialing 7 digit calls to the FXO, but also had a weird double (one after the other) ringback? Also use dtmfmode=RFC2833 in the extension and set the HT-488 the same. I had to give up on that device due to poor audio quality and echo issues . Also intermittent hanging made this device unacceptable for me. Let us know if it works for you? Also which firmware and asterisk version are you using? Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 计划生育的无耻宣传 该结束了
真的很遗憾。不管左派的网友还是右派的网友，在谈到计划生育的时候大都会摆出一副冷酷的面孔。我就来说说计划生育是个什么东西。 坐在电脑面前的精英们应该知道这么一个国情常识：中国的农民是没有任何退休金和任何形式的医疗保障的。 你们有没有想过，他们如果没有一个强有力的孩子，当他们失去劳动能力的时候，就只能坐在家里慢慢饿死？死并不可怕，对于中国农民来说，每年的非正常死亡不计其数：有死在矿井里的，有死在城市的工地上的，有死在收容所里的，有洪水淹死的，有吞农药自杀的，有上访的时候跳楼的，当然也有死在强制堕胎的病床上的。这些都不能让农民恐惧，为什么？因为他们总怀着一线希望能逃过这些磨难，他们一生都以极高的热情在和这些死的可能性做斗争。但是有一种磨难是不可能逃过去的，那就是衰老。 如果一个农民在只有一个女儿的情况下被结扎了，那么就意味着他在很年轻的时候就已经预见到了自己的晚年：除非自杀，否则就只能在极度的物质匮乏中衰竭而死，失去劳动能力的一天就是他们的死期。这种对死亡的确切的预期是多么恐怖你们想过么？一个人在年轻的时候就能预见到自己怎么死，这好玩不好玩呢？ 当然，中国农民的职业寿命也是非常之长的。在网上有很多图片都是70岁以上的老人在背柴火或者乞讨，他们算是很幸运的，自己尚能够"老有所用"养活自己，但是他们很清楚等待他们的将是什么：生命中注定没有一天的假期，退休的日子便是他们的生活来源彻底枯竭的日子。 有人开始从理论上做分析了。即使生的是女儿也有赡养老人的义务啊！就算嫁出去了，她和他老公的财产是双方共有的呀！您要是这么想，就麻烦你到农村看一看吧。农民并不是都不尊重自己的老婆，我也见过感情好的。但是女性对家庭财产的支配权真的是微乎其微，想把种地挣的一点钱拿回娘家去给女方的父母花？看着吧，老公的棍子就要下来了。 我也想过，如果女婿不承担赡养义务是不是可以打官司呢？可是稍微动一下脑子就知道，这根本是不可能的。连给爹妈的钱都没有，难道能有交诉讼费的钱么？我认识一个打过离婚官司的小时工，她曾告诉我，他老公威胁法官：只要你判我离婚，我就砍你全家。当然，我举这个例子决不是想说明基层的法官好欺负，他们可不是善主。只是面对一无所有只有烂命一条的百姓，他们是不愿意拼命的。 中国的农村，想通过法律手段解决问题？想起来脑袋都大了。 应该说，极不健康极不公正的社会环境在逼迫农民生育，而凶残和腐败的地方执法机构又在用各种方式制止农民生育。要减少人口，是应该助长后者还是结束前者呢？或者说，是应该尽量逼迫不生育，还是应该尽量不逼迫生育呢？ 如果真的是为了人民的福利，这个国家有太多比计划生育更可行的方法了。但是这些方法大人们时不屑于用的。 除了增加养老和医疗保障之外，还有一个最简单的办法，就是土地私有。在土地国有的大环境下，农民除了孩子是一无所有的，只有生一个胖小子能让农民找到一点拥有的感觉。 土地国家或集体所有的条件下，有一个永远无法解除的困境，就是土地如何分配的问题。即使我们把农村的官老爷想象得无比清廉，无比公正，那么请问，当人口发生变化的时候，他们如何公正地调整土地使用权呢？ 当然，各地都有不同的办法。我查了各种土地方面的法律，大多语焉不详（原谅我没有做过什么乡土调查，没人给我报路费啊），但是总的来讲，还是以人口为基准的。换句话说，农民多生孩子虽然要被罚款，但是在分配土地使用权的时候，还是会有一些隐性的好处。 而土地私有以后的农民就不一样了，因为有了自己的财产，自己种不动了可以出租，自然心里就塌实了。生多了孩子不仅不能带来什么好处，反而会因为劳动力过剩而降低自家的生存质量。那么不用你计划，人家也自然会去限制生育。 土地私有对于大人们来说当然是不能接受的了。正是靠着对土地的所有权，国家把人民牢固地掌握起来：因为你脚下的土地都是国家的，只要你不会飞，你就时时刻刻地欠着国家的人情，因为你踩了它的地。正因为如此，无论城市还是农村的暴力拆迁都显得那么理直气壮。 计划生育嘛，呵呵，正是这样一种和土地国有相辅相成的政策：国有的土地相当于农场主的一个巨大的畜栏，被限制生育的人民像是被阉割的只能干活的牲畜。这两种措施有效的让人民对国家的依附关系建立得天衣无缝。 以上只是说农民为什么要生的问题。还有就是，人口多究竟有没有那么可怕。有人计算过新增的国民要吃掉多少GDP云云。我听了简直要喷。中国的农民确实是劳动生产率低，这我承认，但是人家什么时候吃过别人创造的财富了？中国农民每年要给国家上缴各种税费，而从来没有得到过一分钱的福利，每修一段破烂公路还都要强行的集一次资！请问，他们消耗掉国家什么了？你们这些白领创造的GDP有哪一分钱是进入了农民的腰包了。不会把你给你家保姆发的工资也算上吧？啊？没人逼你雇保姆啊！ 恰恰相反，超生不但没有给国家带来负担，反而让地方政府有了更好的剥皮抽筋的理由，计划生育官员就像大城市里的交通警和小城市里的扫黄警察一样，每天都在期待着有人犯法，好来送钱给他们。 你们可以去设想未来中国的福利如何如何。但是在这个年代，社会福利对于户口本上写着"农业"二字的人来说还是一个虚拟物品的时候，请不要去咒骂别人占用你的GDP好不好？网上有的是中国底层的照片，你看人家哪个像是吃你们丫的GDP过活的？有的冷酷并不是道德原因造成的，而是因为逻辑思维的缺乏，那就好好锻炼一下你的逻辑思维。 有人提到超生导致的残障人口。避免先天性残障当然是任何一个政府都会做的。但是，我还想提醒一下，中国大部分残障人士也是没有任何福利的呀！也是只能家人养着的呀！即使是享受微弱福利的城市户口的残障人士，他们的数量也远没有中国贪官污吏的数量多吧。而一个乡镇级贪官的开销（包括汽车、手机、吃喝、嫖、旅游、盖办公楼、名牌烟酒、送子女去省城上学……）按一个月5000块算不多吧？那就顶得上20个城市贫民的最低生活保障（也就是国家花在他们身上的所有的钱）。至于县级？市级？省级？X级……的干部，一人顶1000个残疾人不在话下吧？ 计划生育和反贪也许并不截然矛盾。但是把计划生育上升到基本国策，分明就是把国家落后的责任推卸给普通老百姓。如果有这么一个人，他在声色场所挥霍无度，却在去菜市场买菜的时候讨价还价，你会不会觉得他有病？国家花那么多力量来搞计划生育，正是这样一个有病的表现。 当然，中国经常干这种事情。比如希望工程吧，这么多年据说也就募到了20个亿。你说好笑不好笑，国家随便少干一件蠢事不能省出20个亿？要让我们捐钱？为什么要丢西瓜拣芝麻，这可能只有政策制定者自己心里清楚。要不大家都来猜一猜？ 然后，请允许我再往下说一层。 人到底是什么？是一个国家富强的手段，还是一个国家富强的目的？人口问题？人口不是问题，人口不就是你和我构成的？人口不是国家豢养的牲口，需要耕地或挤奶就多产一点，养活不了就少产一点。恰恰相反，人口是这个国家的主人，国家要无条件服从人口的需要而不是相反。 如果一个国家的妇女要承受强迫结扎、强迫堕胎的痛苦，要被别人用暴力剥夺自己腹中的胎儿，这个国家再富强又有什么意义？当妇女们被成群关在拘留所里，警察等着她们一个个地签字同意结扎，然后直接用卡车拖到医院，这个国家作为一个人类生活的地方还值得存在下去么？ 中国妇女当然从来没有过过好日子：一夫多妻、裹小脚、用生命保贞洁……但也从没有像现在这样被剥夺了亚当夏娃时就有的伟大的生育的权力呀。 正是因为用考虑畜牧业的方式来考虑人口，把农民当成国家的财产而不是主人，才会出现这样一个荒谬的情况：一方面总说人口多，一方面却无耻地限制老百姓出境，对于基层老百姓办护照百般刁难！ 不是说人多么？为什么不让人家到别的国家去？为什么办护照还要审批？为什么北京上海这些所谓的"高素质"人口出国反而不受限制，为什么农民跑出去就不行？如果不是把人家当成田地里的劳动机器，还有什么其他原因呢？ 我想请大家看一条很少被注意的法律。这是《中华人民共和国出入境管理法实施细则》中的一句话："出境就业，须提交聘请、雇用单位或者雇主的聘用，雇用证明"。这里的"提交"不是向负责签证的老外提交，而是向"户口所在地的市、县公安局出入境管理部门"提交。如果按照某些大人们抱怨的那样，中国穷是因为人太多，那应该积极鼓励大家出国打工才好。当然不要求领导们花时间去帮他们在国外找工作，但至少不该限制人家。即使没有"雇佣证明"，人家出去以后再想办法又有什么关系呢？ 别告诉我什么"给国家丢脸"。让贫苦的农民担负起给国家挣面子的责任是毫无道理的。请问他们在这几十年的生活里，什么时候有过尊严可言？不能让占人口大多数的农民过得高兴，这个国家还能有面子么？ 如果有人提出通过饿死一批人来减少人口，大家肯定不会同意。因为你们都知道生存权是全世界公认的人权，甚至中国还把它说成是"中国对人权理论的一大贡献"。但是通过限制生育甚至强迫结扎来减少人口，大家居然就认可了，也就是说，一般人认为生育权没有生存权那么重要。 可是，你们知道么？对于任何一种正常的生物来说，生育都是比生存更加神圣的使命。人也是不能例外的。对于没有宗教的民族尤其如此，因为只有基因的延续能给人带来永恒体验。 我们来举一个例子。设想一个母亲有不止一个孩子，当其中一个孩子的生命受到威胁的时候，你说她会不会用自己的生命去换取这个孩子的生命呢？我可以告诉你们，99%的母亲都会这样的。为什么呢？这是所有能在进化大潮中保留下来的基因共有的自我保护机制在起作用，它们在下意识中暗示着每个人：牺牲个体，让基因延续下去。 不排除有能生育而不愿生育的人，就好像有能活下去但选择自杀的人一样。这是另一回事。我现在说的是，对于想生育的人不允许其生育是多么的残忍。 凡是为计划生育基本国策叫好的人，请你们务必发发慈悲，看看农民的生活现状。以这样奴隶般的生活质量，即使是纯粹为了高兴而生孩子也是毫不过分的。 这个国家欠农民的太多了，看在1960年前后那3000万冤魂的份上，别再折磨他们了吧。 -- Jefferyiaxtel Num: 1-700-576-1311fwdnet Num: 728150 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
