Re: [asterisk-users] agi with java?
Hi Lee, we are a Java shop and our experience with Java has been much the one you say - it does scale pretty well and it is very solid. What I was trying to say is that Java is not very well suited to the classic, Unix-style, fire-up-process-and-let-it-die that goes for CGI/AGI programming. On the other side, I have no doubt that with an application server and FastAGI you can get quite a lot of bang for the buck. :) l. On Fri, 08 Jun 2007 18:07:50 +0200, Lee Jenkins <[EMAIL PROTECTED]> wrote: We have found that generally speaking, running the FastAGI server on the same machine as Asterisk yields better performance than launching separate exe processes through the dial plan. Completely anecdotal of course. This is careful research conducted over our entire 5 customer base... -- Loway Research - Home of QueueMetrics http://queuemetrics.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
Re: [asterisk-users] IAX trunk with dynamic IPs
In your no-ip client set it to update ip every 2 minutes or so . and /etc/asterisk/dnsmgr.conf set refresh interval as 30-40 by defualt its 300 ( 5 minutes) On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote: Hi Matt, Every time I do that, IAX stop sending the POKE messages (necessary for trunk management). Do you know what could be happening? Thanks. Ronaldo. Matt wrote: > > *set "enable=yes" in the "[general]" section of > /etc/asterisk/dnsmgr.conf* > > > > > ___ > --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] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event
Lee Jenkins wrote: Hi all, My company has pretty much standardized on Polycom phones and I am in the beginning phase of writing a GUI for administering/managing polycom provisioning at multiple sites which we intend to release as OS. Good luck. I've noticed that Polycom makes changes to the spec on both minor and major releases. I don't know how you will account for this, but as the saying goes; "Damn it Jim! I an admin not a developer!" voIpProt.SIP.requestValidation.x.request.y.event I understand what it does (at least conceptually) but ss the "x" variable still referring to a server (1 or 2)? And the "y" var, what is it referring to? An event? Which one? I believe that the x and y are counters used to group attributes together. The attribute above; voIpProt.SIP.requestValidation.x.request.y.event, depends on voIpProt.SIP.requestValidation.x.request being set to either "SUBSCRIBE" or "NOTIFY". Since you SUBSCRIBE to specific events, this is a way for the phone to validate either all events, or just those you specify. I have not tested the following examples, and welcome any correction from someone who has, but this is my understanding of the settings. The default; Could be; Or possibly; voIpProt.SIP.requestValidation.1.request.2.event="check-sync"> I believe that you can validate that a check-sync event came from the server and not from some other system on the LAN by: Please let me know if you discover that I'm wrong about this, but I believe that is the way it's supposed to work. Jeff -- Jeff Davis Netsource Consulting ___ --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] IAX trunk with dynamic IPs
Hi Jaswinder, That is what I did. The thing now is, when I set "enable=yes" in /etc/asterisk/dnsmgr, Asterisk stops sending IAX POKE messages to the remote peer (to keep the trunk up). I've searched on the Internet but I couldn't find any documentation about how "DNS update manager" works for Asterisk. Do you have any? Ronaldo. Jaswinder Singh wrote: In your no-ip client set it to update ip every 2 minutes or so . and /etc/asterisk/dnsmgr.conf set refresh interval as 30-40 by defualt its 300 ( 5 minutes) On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote: Hi Matt, Every time I do that, IAX stop sending the POKE messages (necessary for trunk management). Do you know what could be happening? Thanks. Ronaldo. Matt wrote: > > *set "enable=yes" in the "[general]" section of > /etc/asterisk/dnsmgr.conf* > > > > > ___ > --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
[asterisk-users] IAX Peers show command
Hi all, What does (T) mean on the output of "iax2 show peers"? The following my output. darkstar*CLI> iax2 show peers Name/UsernameHost Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0 UNKNOWN sp/ata201.26.67.102 (S) 255.255.255.255 4569 (T) UNKNOWN 2 iax2 peers [0 online, 2 offline, 0 unmonitored] Ronaldo. ___ --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] agi with java?
On 10 Jun 2007, at 13:29, Lenz wrote: Hi Lee, we are a Java shop and our experience with Java has been much the one you say - it does scale pretty well and it is very solid. What I was trying to say is that Java is not very well suited to the classic, Unix-style, fire-up-process-and-let-it-die that goes for CGI/AGI programming. On the other side, I have no doubt that with an application server and FastAGI you can get quite a lot of bang for the buck. :) l. Just to encourage folks who run a mile from the whole J2EE bloated mess.. You can write a very simple FastAGI server using asterisk-java in J2SE, and run it in just about any JVM. We have been very happy with it. Tim. ___ --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] Blocking 900 calls
Presently I have _all_ 900 calls blocked in Asterisk 1.25 but today I had to call a parts vendor at a 972 number. What are the "safe" 900 numbers - meaning the ones that are not sex lines that change by the minute? Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blocking 900 calls
Presently I have _all_ 900 calls blocked in Asterisk 1.25 but today I had to call a parts vendor at a 972 number. Blocking anything with 9XX isn't a good idea. There are lots of regular area codes in the 9XX block -- take a look: http://www.localcallingguide.com/lca_listnpa.php?section=9 I *think* only the exact 1-900 prefix is a premium rate call. --Luki ___ --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] Blocking 900 calls
Luki wrote: Presently I have _all_ 900 calls blocked in Asterisk 1.25 but today I had to call a parts vendor at a 972 number. Blocking anything with 9XX isn't a good idea. There are lots of regular area codes in the 9XX block -- take a look: http://www.localcallingguide.com/lca_listnpa.php?section=9 I *think* only the exact 1-900 prefix is a premium rate call. --Luki http://en.wikipedia.org/wiki/Premium-rate_telephone_number ___ --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] Hot GXP-2000
Yikes!
{sarcasm on}
Yea, why use good stuff when you can get stuff at less than half the price.
Who cares if it ACTUALLY works properly. As long as it's cheap.
{/sarcasm off}
Yikes!
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Saturday, June 09, 2007 10:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hot GXP-2000
On 6/10/07, Bill Hackensack <[EMAIL PROTECTED]> wrote:
> Why? Because of their excellent customer support in taking care of a
> problem? At the price of the Grandstreams compared to others, I can
> deal with a couple of bad apples. I can buy two Grandstream's for the
> price of a phone with similar features. I can deal with a lot of bad
> apples at that ratio. Their sidecar is so cheap compared to others it's
not even funny.
>
> Plus, I can't even get some of the functionality the GXP's give me
> from other phones.
However the problem is it's not even a phone.
>
>
> On 6/9/07, Dovid B <[EMAIL PROTECTED]> wrote:
> > One of the reasons why I stand clear of Grandstream
> > - Original Message -
> > From: "Carlos Chavez" < [EMAIL PROTECTED]>
> > To: "Asterisk"
> > Sent: Friday, June 08, 2007 6:47 PM
> > Subject: [asterisk-users] Hot GXP-2000
> >
> >
> > > ___
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> > >
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> >
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> > asterisk-users mailing list
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> >
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> >
>
>
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Re: [asterisk-users] IAX trunk with dynamic IPs
Hi Ronaldo - I have a IAX trunk between two asterisk servers, both with dynamic IP and both have a DNS name associated with it. In the iax.conf file I configure the "host" parameter with the DNS name of the servers. Everything works fine until one of these servers get a new IP, so the other can't find its peer (the one that has just gotten a new IP). If I manually issue a "iax2 reload" in the CLI, asterisk tries to find the IP of the peer (based on its DNS name) and everything starts working again. This is the section for my trunk in one of my servers: [sometrunk] type=friend username=someusername secret=somesecret auth=plaintext host=host.no-ip.org context=incoming peercontext=incoming qualify=yes trunk=yes Is there any way to tell asterisk to try to find the peer's IP address if that peer is "unreachable" or each 10 minutes? I don't know if your DDNS provider would support this, but if you set the TTL value of your DNS hostnames to something very low, like 10 seconds, it would force your OS to keep finding the latest IP. - Noah ___ --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] Bad Echo between SIP calls
Deepak Naidu wrote: Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to PRI we were till date using 4 analog lines connected with TDM card from digium & no echo for pure SIP to SIP lines. Now I have TE212P which had onboard echo cancellor. I am trying make myself clear before I blame on any network. B'cos for sure we have a spegati of networks & no QoS. Also the intresting thing is if I call from one extension to other dialing the main line & then extension the call is crystal clear. but when dialing a direct extension its a hell of echo. Make SURE you have the handset plugged into the handset port of the phone, not the headset port of the phone. ___ --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] IAX trunk with dynamic IPs
Hello You should use qualify=310 ( any value in millisec ) .. qualify=yes is not proper . I am not sure about how asterisk's dnsmgr manages dns refreshing but maybe someone else can answer that question . On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote: Hi Jaswinder, That is what I did. The thing now is, when I set "enable=yes" in /etc/asterisk/dnsmgr, Asterisk stops sending IAX POKE messages to the remote peer (to keep the trunk up). I've searched on the Internet but I couldn't find any documentation about how "DNS update manager" works for Asterisk. Do you have any? Ronaldo. Jaswinder Singh wrote: > In your no-ip client set it to update ip every 2 minutes or so . and > /etc/asterisk/dnsmgr.conf set refresh interval as 30-40 by defualt > its 300 ( 5 minutes) > > On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote: >> Hi Matt, >> >> Every time I do that, IAX stop sending the POKE messages (necessary for >> trunk management). >> Do you know what could be happening? >> >> Thanks. >> Ronaldo. >> >> Matt wrote: >> > >> > *set "enable=yes" in the "[general]" section of >> > /etc/asterisk/dnsmgr.conf* >> > >> > >> > >> >> > >> > ___ >> > --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] ast_dynamic_str_thread_build_va() is defined with 6 args but only called with 5 args??
va_list is a macro used to define optional arguments, so I suppose is correct to call it with 5 or more arguments. I suggest you to compile asterisk without optimizations ( search in voip-info.org ) and then open a bug ( make sure first you read the bug guidelines ) in bugs.digium.com Regards On 6/9/07, Frank Tarczynski <[EMAIL PROTECTED]> wrote: I'm having a problem with asterisk-1.4.4 dumping core under Solaris 10 with a SIGSEGV error. gdb gives this stack trace: #0 0xfebd4d0c in strlen () from /usr/lib/libc.so.1 #1 0xfec2a386 in _ndoprnt () from /usr/lib/libc.so.1 #2 0xfec2d4bb in vsnprintf () from /usr/lib/libc.so.1 #3 0x080e86de in ast_dynamic_str_thread_build_va (buf=0x8172763, max_len=0, ts=0x81482a0, append=0, fmt=0x811dc6a "%25.25s %s\n", ap=0x8046f18 "X'\027\b") at utils.c:969 #4 0x080890e0 in ast_cli (fd=1, fmt=0x811dc6a "%25.25s %s\n") at cli.c:69 #5 0x0808c946 in help1 (fd=1, match=0x0, locked=0) at cli.c:1746 #6 0x0808ca67 in handle_help (fd=1, argc=0, argv=0x8047080) at cli.c:1773 #7 0x0808d664 in ast_cli_command (fd=1, s=0x0) at cli.c:1979 #8 0x08073d0f in main (argc=135695685, argv=0x80471f4) at asterisk.c:1384 I've noticed that ast_dynamic_str_thread_build_va is defined in utils.c on line 969: int ast_dynamic_str_thread_build_va(struct ast_dynamic_str **buf, size_t max_len, struct ast_threadstorage *ts, int append, const char *fmt, va_list ap) and it's called in cli.c on line 69: res = ast_dynamic_str_thread_set_va(&buf, 0, &ast_cli_buf, fmt, ap); Most interesting is that the function is defined with 6 arguments and only appears to be called with 5(?). Is this correct? Frank ___ --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 -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.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
[asterisk-users] best format for audio via asterisk...
Hi all - We are using voicepulse connect & asterisk together. We'd like to record our own outgoing messages, to be played back people that will be dialing into our voicepulse connect supplied DID. (am I getting this lingo right?) What is the best audio format to ensure the highest level of audio quality? Thank you - Matt Pease Parking Hero ___ --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 sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event
Jeff Davis wrote: Lee Jenkins wrote: Hi all, My company has pretty much standardized on Polycom phones and I am in the beginning phase of writing a GUI for administering/managing polycom provisioning at multiple sites which we intend to release as OS. Good luck. I've noticed that Polycom makes changes to the spec on both minor and major releases. I don't know how you will account for this, but as the saying goes; "Damn it Jim! I an admin not a developer!" voIpProt.SIP.requestValidation.x.request.y.event I understand what it does (at least conceptually) but ss the "x" variable still referring to a server (1 or 2)? And the "y" var, what is it referring to? An event? Which one? I believe that the x and y are counters used to group attributes together. The attribute above; voIpProt.SIP.requestValidation.x.request.y.event, depends on voIpProt.SIP.requestValidation.x.request being set to either "SUBSCRIBE" or "NOTIFY". Since you SUBSCRIBE to specific events, this is a way for the phone to validate either all events, or just those you specify. I have not tested the following examples, and welcome any correction from someone who has, but this is my understanding of the settings. The default; Could be; Or possibly; voIpProt.SIP.requestValidation.1.request.2.event="check-sync"> I believe that you can validate that a check-sync event came from the server and not from some other system on the LAN by: Please let me know if you discover that I'm wrong about this, but I believe that is the way it's supposed to work. Jeff Thanks Jeff. -- Warm Regards, Lee ___ --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 problem...
Thankyou to all of you who replied. Seing as how this seemes to be an un-implemented feature under *, I will go ahead and write a handler for it. I will post here as I have progress. -Chris A side note: The reason I am doing this, is I do computer repair, I have brodband at my office, and have an * box doing my phones. I want to be able to "dial-in" with computers I am working on to test and make sure they are working, but I dont want to pay for a dialup account. I could order a line just for this, but it is something I do so rarly, that it would be more economical to just write a handler for *. ___ --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: Asterisk RAS
(Sorry forgot to chaange the subject) Thankyou to all of you who replied. Seing as how this seemes to be an un-implemented feature under *, I will go ahead and write a handler for it. I will post here as I have progress. -Chris A side note: The reason I am doing this, is I do computer repair, I have brodband at my office, and have an * box doing my phones. I want to be able to "dial-in" with computers I am working on to test and make sure they are working, but I dont want to pay for a dialup account. I could order a line just for this, but it is something I do so rarly, that it would be more economical to just write a handler for *. ___ --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] basic asterisk knowledge
I have question concerns asterisk 1-What is difference between G.729 and G.729A? 2-How can I know the requirement hardware for 150 extension on asterisk 1.4.4 making 50 simultaneous call? 3-Do asterisk have a codec conversion? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Going to VON Stockholm? Meet you at the Asterisk BOF!
The Asterisk BOF session will be tomorrow at 4.30 PM, VON Stockholm at the Stockholm Fairgrounds in Älvsjö. http://tinyurl.com/3degv5 From time to time you will find me in the Voop stand in the Digium/ Asterisk Pavillion in the exhibition. See you! /Olle - Sponsor Codename Pineapple - the Chan_sip3 development! http://www.codename-pineapple.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
