[asterisk-users] Jitter buffer not used in SIP - chan_local - ZAP path even with /nj for local channels
Hi, Asterisk 1.4 Working (jitter buffers created as expected): ZAP - SIP SIP - ZAP Not working (no jitter buffers created): SIP - chan_local (with /nj) - ZAP SIP - chan_local (with /j) - ZAP SIP - chan_local (with no flags) - ZAP I have this in zapata.conf: jbenable=yes jbforce=no jbimpl=fixed jbmaxsize=300 Is there something I haven't tried that will make this work or will I have to change my dialplan so it doesn't use local channels? Thanks, Mike PS, here are some pages that I have used as sources of information: No mention of /j for local channels http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels Nothing about local channels http://www.asterisk.org/node/48317 Mentions /j for local channels to apps http://www.voip-info.org/tiki-index.php?page=Asterisk+new+jitterbuffer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Have problem with realtime sql
That's from asterisk-addons, you can ignore that error. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mark morreny Sent: Tuesday, March 25, 2008 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Have problem with realtime sql Hi, I am having a strange problem with attempting to get voicemail-to-mysql to work. The biggest problem is that I am not able to store voicemail into database. So, I followed the instructor found on the web: Updated the /usr/src/asterisk/apps/Makefile to have USE_MYSQL_VM_INTERFACE=1 and recompiled asterisk, with make clean; make; make install (By the way, is it necessary to update the Makefile for Asterisk 1.4.18?) After make install, I got some warning messages: Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so chan_ooh323.so format_mp3.so res_config_mysql.so Is this the problem that causing Asterisk not able to store voicemessages to mysql? If so, how do I fix it? From the console, I can get realtime status ok: CLIrealtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username askuser for 1 minutes, 34 seconds. Thank you very much for your kind attentino. You help is greatly appreciated. Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip exten matching based on contact: sip header?
Asterisk: 1.4.17 with sip realtime Openser 1.3.x Hi, I had this setup working fine until I try putting OpenSER in the picture as a proxy. Unauthenticated calls go to a PRI based app via a ZAP channel, calls to sip users get send to them etc. Now with a proxy in the picture asterisk asks the incoming calls for authentication 407 Proxy Authentication Required. It seems that the sip channel matching is based only on source IP address instead of also checking the contact: header as mentioned in the O'Reilly book. According to Asterisk 2nd edition it says about insecure ... If you set insecure=invite, you'll determine which peer to match on by comparing the IP address or hostname and port number to those provided in the contact field of the SIP header with the host and port options in sip.conf. If a match is found, authentication will not be required on the initial INVITE, and the call will be allowed. The funny thing is that if I do a 'sip reload' and receive a call from one my DIDs through the provider it goes to the default context when received through OpenSER as expected. But once a sip realtime user makes a call it will match their peer instead of the one specified with the provider's Ip address. I've seen this in my logs after turning on sip debugging, it looks like different users get matched based on the sort of the sip peers list (which can change based on how long ago a reload was done and who has been active because of sip realtime). [Mar 24 17:04:23] Sending to 74.x.x.x : 5060 (no NAT) [Mar 24 17:04:23] Using INVITE request as basis request - [EMAIL PROTECTED] [Mar 24 17:04:23] Found peer 'some_peer' The sip users have their host=ip_of_openser so I can understand why it would get confused if it didn't check the contact header for clairification since a call is also coming from that source IP address when proxied through openser. Maybe I'm approaching this from the wrong direction, anyone have any ideas? Mike [privider1a] type=peer host=67.x.x.x insecure=invite,port context=default qualify=999 [provider1a] type=peer host=67.x.x.x insecure=invite,port context=default qualify=999 [provider2] type=peer ;host=sip.provider2.com host=64.x.x.x insecure=invite,port context=default qualify=999 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Subexpression usage in Asterisk Dialplan Regular Expressions
Hi,
I currently have these two lines in my dialplan to extract different parts
out of a variable and I'd like to do it in one line instead. Does anyone
know how to use regular expression subexpressions in the dialplan?
Outputting a comma separated list that can be sent to ARRAY() would be nice
too (tried that, didn't work -- only got the first subexpression).
;extract dialed number
exten = s,n,Set(dialed_num=$[ ${ARG1} =~ (.*)\\* ])
;extract user specified callerid
exten = s,n,Set(callerid_num_custom=$[ ${ARG1} =~ \\*(.*) ])
Mike
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[asterisk-users] Problem sending CallerID Name to Dialogic based phone app
Hi, Asterisk 1.4.17 Sangoma a102DE I'm having some issues sending CallerID Name to a Dialogic based phone app. According to the pri debug (asterisk2a-pri-debug.txt in [3]) you can see that it is sending the CallerID Name Mike - Budgetone - reachme.com to the Dialogic card, but it isn't regestering on the Dialogic based system. I can receive CallerID Names from our Paetec, our provider on the Dialogic system, so it must be using a different PRI signalling message. Right now, the Sangoma card is not connected to our provider, only to the Dialogic so getting a PRI trace from asterisk would require doing a trace on the dialogic (which will probably give different debug format) or connect the provider directly to sangoma/asterisk which would require *a lot* of painful configuration changes, mostly on the Dialogic side. I'm hoping someone can help shed some light on this. I simply want to get asterisk/sangoma to send the same signalling that Paetec does. I am contacting them for details on the setup they are using also. http://www.dialogic.com/manuals/docs/isdn_api_v5.pdf The Dialogic system is using the U_IES[1] with cc_GetCallInfo() to get the information sent from Paetec, our provider, but the information isn't being sent the same way from the Sangoma card, even though I have it set to pri_net in zapata.conf. Alternatively, we can check the UUI[2] message to see if the Dialogic card will receive it that way. Mike 1. Information Elements (IEs) in CCITT format. The cc_GetCallInfo( ) function retrieves all unprocessed IEs in CCITT format. Be sure to allocate enough memory (up to 256 bytes) to hold the retrieved IEs. The IEs are returned as raw data and must be parsed and interpreted by the application. Use IE_BLK to retrieve the unprocessed IEs. For a description of the IE_BLK data structure, see Section 6.6. IE_BLK. See Appendix C for descriptions of information elements specific to the DPNSS protocol. 2. User-to-user information. The user information data returned is application-dependent. The user information is retrieved using the USRINFO_ELEM data structure. For a description of the return format for UUI, see Section 6.16. USRINFO_ELEM. 3. http://reachme.com/sangoma-asterisk-dialogic/sangoma1a-scripted-config-info. txt http://reachme.com/sangoma-asterisk-dialogic/sangoma1b-wanrouter-restart.txt http://reachme.com/sangoma-asterisk-dialogic/asterisk2a-pri-debug.txt http://reachme.com/sangoma-asterisk-dialogic/sangoma2b-pri-show-span-x.txt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does the meetme module still require anexternal timing source?
Agreed, Callweaver and Freeswitch are both better for conferencing (especially if you don't have zaptel hardware). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, March 12, 2008 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] does the meetme module still require anexternal timing source? Try Callweaver. Thanks, Steve Totaro On Wed, Mar 12, 2008 at 4:12 PM, Dennis Christopher [EMAIL PROTECTED] wrote: Thanks Matt, However I am looking to see if Asterisk with meetme is viable on OS X, and I believe that ztdummy will not compile on that platform. If so, I would need an alternative to meetme to do conferencing...? Dennis On 12-Mar-08, at 4:02 PM, Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dennis Christopher wrote: All, Can anyone confirm if the meetme module still requires an external timing source, such as a card and or driver? Correct, but insofar as a driver, you can just use ztdummy, which will be loaded by default when starting up zaptel if you have no hardware installed. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH2DbpDQNt8rg0Kp4RArouAKCF0D36feiSxokdOx8UzF2gGOhonACgou4K WIAhdj/PUrOx5Z4N0fePRqM= =xfLA -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Druid Open Source Edition
I believe that is/was one of the goals of the phonecall project. -Original Message- Does it implement the ability to run more than 1 PBX in asterisk ? (Virtual PBX) To be clear: more then 1 company using the same physical asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk out of service
You'll need to post more info. Version and a scenario of what was happening at the time would be a good start... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rilawich Ango Sent: Wednesday, March 12, 2008 6:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk out of service Hi all, I got the following message in the log yesterday. After that, no more in/out bound call can be made. What is the meaning of the message? ango [Mar 12 09:26:17] ERROR[29565] chan_sip.c: We could NOT get the channel lock for SIP/2367-d8062fb0! [Mar 12 09:26:17] ERROR[29565] chan_sip.c: SIP transaction failed: [EMAIL PROTECTED] [Mar 12 09:33:15] ERROR[29565] chan_sip.c: We could NOT get the channel lock for SIP/2327-dc32e4a0! [Mar 12 09:33:15] ERROR[29565] chan_sip.c: SIP transaction failed: [EMAIL PROTECTED] [Mar 12 09:35:19] ERROR[29565] chan_sip.c: We could NOT get the channel lock for SIP/10.201.2.224-0e914380! [Mar 12 09:35:19] ERROR[29565] chan_sip.c: SIP transaction failed: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended FXO device
Rich Adamson wrote: I've tested a large number of other external adapters and have not found a single one that had a reasonable echo canceller built in. Many of them work fine on short pstn lines (where echo is much less of a problem), but provided even reasonable service on longer pstn lines or lines that involve unusual telco configurations (eg, remote line concentrators). What about devices from audiocodes, ipgear/boscom and vegastream? Can you give a list of products you have tested and your results as well as your testing environment and methodology? ___ --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
Octasic for TDM2400P and TDM400P? was: [Asterisk-Users] TE420P/TE415P?
When will Digium include the octasic on the TDM2400P? And maybe the TDM400P? Also how does the TE415P and TE420P differ from the TE412P card? ___ --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] Wiki Voip Phone reviews
Hi, We have a page on the wiki just for phone reviews, but I think it needs a bit of format change. Instead of individual reviews for each phone, I think each person should review all phones they have worked with and list the phones they have had access to and rank them in relation to each other. Also each review should have a date so the reader can see how fresh the data is to current. http://www.voip-info.org/wiki/view/VOIP+Phones+Reviews An example would be: June 28th, 2006 Mike Fedyk I have used these phones and I rank them in this order: Linksys 941 Polycom 301 Sipura 841 Grandstream GXP-2000 Linksys SPA-941: Pros: Has a feature set of a much more expensive phone. Well laid out menus and buttons. 2.5MM plug for headset. Very intuitive interface. Cons: The screen resets whenever the phone checks for updated config, (could also be caused by sip notify messages also) The blended line appearance feature requires SIP-B and Asterisk won't be supporting that soon. Comment: The handset microphone is *very* sensitive -- it will pick up the entire room sounds and conversations like a speaker phone. You should consider reducing the microphone gain in the config. Like the SPA-841 This is the phone that I would choose for myself (though I want a 942). Polycom IP301: Pros: Very nice buttons, high sound quality handset and speaker phone, separate button for headset, nice bright flashing message waiting indicator (MWI) Cons: Speaker phone is one way, you have to use the handset or headset if you want to speak and that deactivates the speaker phone. The menu layout is terrible. For instance, if you want to get to the recently called list, you have to press 5 buttons. Compared to one button accessibility to the same list on Sipura/Linksys phone and you will notice the increased amount of time you spend on the polycom menu system. Comment: The headset jack is a RJ-11 used by your standard business headset companies like Plantronics and others. That can be a benefit depending on whether you already have the headset amplifiers or not. This is what sits on my desk now. Though not for much longer. Sipura SPA-841: Pros: Low cost, works well, very intuitive interface. Cons: Discontinued by Linksys. Hard to read screen, low quality screen, screen at bad angle, very light and small so pressing keys usually moves whole phone, buttons are rubbery and feel as if they would stick but don't, speaker phone could be better. Comments: Like the SPA-941, once you start using the phone the buttons and menus are very intuitive. You won't have to read the screen, you just press a few buttons in a row and it does what you want. Grandstream GXP-2000: Pros: Backlit screen, firmware getting better, big buttons. Cons: Very long list, see gxp-2000 page. Poorly tested and buggy firmware. Comments: I have a client that needed phones to connect to vicidial and got these. I don't recommend them for much else but the backlit screen on a cheap phone. Beware. I'll spend more time on the GXP-2000 entry when I modify this post for the wiki, this message has been in writing long enough already. Mike ___ --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] Mail loop?
Is anyone else getting messages from the lists.digium.com mail server with errors about a mail loop? I've been getting this for the last few weeks, but I don't have any list software on my server. Any ideas? ___ --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] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. I don't know Portuguese and my Spanish is terrible, but I understood that Josue wanted to know if he needed any external modules. Marco pointed him to the right place to get skype-to-sip and now they're going to collaborate. So, please guys English please or you'll get more of my bad translations. ;) Mike ___ --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] GXP-2000
Kristian Kielhofner wrote: Mike Fedyk wrote: I happen to have asterisk running as a router, so I use it doing QoS with tc (traffic control) and wondershaper set to prioritize based on port ranges. I sent a patch to the debian bug tracking system a while back with a few improvements -- I should check on that. It basically prioritizes smaller packets before larger packets with ~8 levels of priority and groups of sizes for the packets. Just doing that automatically handles 80% of the need for prioritization without specifying port ranges for the sip rtp packets. Mike Mike, Have you tried AstShape? Shapping based on port ranges is totally hit or miss. TOS is the way to go: http://www.krisk.org/files/astlinux-i586/usr/sbin/astshape Comment out the . /etc/rc.conf and you should be okay! Actually the above is wrong. I don't use port ranges at all, just packet sizes. It allows me to blast away with p2p, interactive ssh and scp file copies all while having two g.711 and one g.729 voip conversation going on a dsn connection with a 384Kbps upload speed. It is based on the premise that smaller packets should have higher priority. There will be exceptions of course, and empty classes have are there for that also. For the common case, no configuration is necessary. Give this one a try: http://mikefedyk.com/wondershaper-pkt-size-classes Mike ___ --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] finding mac addresses
arp in the shell mojowrkn wrote: All, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools. Thanks-John ___ --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] Echo Problem with T411P
Kevin P. Fleming wrote: - Steve Davies [EMAIL PROTECTED] wrote: :) Now you've defeated me. I imagine that you need to do something to enable EC on that card, but it is not a card I know, so I'll leave it to someone who knows the card to offer any suggestions. The only requirement is that 'echocancel=yes' is present in zapata.conf for those channels. If the hardware echo canceler is present and enabled, then it will be used instead of the software canceler for those channels. How can you detect if the HW echo can is enabled? Is it console output during module load or something else? ___ --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] DTMF Talk off
this does not make any sense.
How do you dial to a service provider from your * box? Does it use PPP
and IP? And then you connect to another * box that is on a cable
connection that receives the call over IP and then dials out to a voip
provider? How do any fxo devices come into this picture? How does a
zap channel come into this picture?
John Millican wrote:
Doug,
The interface that i dial to is at my Service provider and am not sure what
they are using. I dial out of my * box to a service provider number which is
answerd by an asterisk box that I have at another location on a high speed
cable connection, that box then dials the numberI ultimately want to reach.
I use an extensions.conf line at my home * such as:
Dial(zap/1/my_sip_numberww${EXTEN});
this works great and saves me a ton on call costs.
John
On Monday June 19 2006 12:19 pm, Doug Crompton wrote:
John,
You said you were using a PAP2.. what is the FXO interface at the (far)
asterisk end?
Doug
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
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Re: [Asterisk-Users] free sun boxes
I'm in southern California, are you close or can you ship? Bob Knight wrote: I have 4 sparc based sun boxes I am about to pay money so I can get rid of them. They are running older versions of Solaris. You should be able to load Solaris 10 and play around with * on them. Time to clean the office: 3 Ultra 5 1 Sparcstation 5 I also have a box full of Sun keyboards and mice. Contact me offline if you want them. I've had many good years of development on them and it kills me to just toss them, but the office is just too damn cluttered. thanks, bk... ___ --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] What ever happened to the LTAPI, the Linux Telephony API?
Hi, I've just been going through the various modules that are autoloaded to see what I need and what I don't and came across chan_phone.so which loads /etc/asterisk/phone.conf. I did a lookup on voip-info and google and came across this article in Linux Journal from 2001. Anyone know why it isn't being used much (from what I can tell) and what's happening with it today? Thanks, Mike http://www.linuxjournal.com/article/4468 ___ --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] Best $300 VoIP phone for asterisk?
Michael Graves wrote: I have the IP600 and like it a lot. However, I really LOVE the Aastra 480i CT. It supports more lines than the ip600, has a backlit LCD, and the cordless handset is GREAT! How is the range, and in what environment did you test? Can you a call on the cordless and the base-station without any loss of functionality (forwarding, 3-way, etc) on either base-station or cordless? Any quirks you've noticed with this phone? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EC needed in all-digital situation?
Warren wrote: So the next question becomes... Is hardware EC necessary or can * handle it in software? I am looking at some pretty beefy hardware for my platform, a Dell PE2850 with dual Xeon 3Ghz processors and plenty of RAM to spare. Can your processors handle the load, yes. Do you want to use software echo cancellation? Most likely no. Should you get a Digium card with hardware echo cancellation? That's debatable. For ~$300 difference the Sangoma four port DS1 card w/ echo can gives you 1024 taps of echo cancellation. In comparison the Digium TE411P w/ echo can gives you 512 taps with 32 or fewer channels active. Once you go over that, you get 128 taps. Digium's hardware echo cancellation provides 64ms across 32 channels; however, when it scales over 32 channels it is reduce to 16ms per channel across all channels. http://blog.voipsupply.com/asterisk_hardware/ If you want to support Digium/Asterisk, then buy a support contract. Admittedly I haven't worked with Digium's DS1 cards, but I have used their TDM400p cards. I can tell you that the software echo can isn't that good. Of course listen to others who are using Digium's hardware echo can. In fact, I'd like to hear from people who have had good experiences with it (and especially if you have bad experiences with sangoma hardware echo can). Until Digium's products have an echo can with a longer tail, buy Sangoma. ___ --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] Best $300 VoIP phone for asterisk?
Andrew Kohlsmith wrote: Again, good to know. Thank you for your detailed post! The XML config for these phones gives them a leg up over the ip501 as well, that is for sure. I believe the IP501 phones do have a XML config file. At least the IP301 does. I take it that you mean the XML is better on the 480i CT. If so, can you be more specific? ___ --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] Best $300 VoIP phone for asterisk?
Andrew Kohlsmith wrote: On Friday 16 June 2006 14:50, Mike Fedyk wrote: I take it that you mean the XML is better on the 480i CT. If so, can you be more specific? No, I mean the XML config file for controlling the screen on the Aastra 480i. There is no such thing on the ip301/501. The ip601 has the minibrowser but I haven't played with that. You mean the menus aren't bass ackwards like the IP301 and you can change them in the XML file? Sign me up! ___ --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] Echo Problem with T411P
Steve Davies wrote: On 6/15/06, Mike Fedyk [EMAIL PROTECTED] wrote: Steve Davies wrote: We have even experienced problems within Europe where providers route national calls via international routes to save money. This adds significant latency and makes any echo so heavily delayed that asterisk cannot remove it. More than 128ms? 128 = 128 taps = 16ms of 8KHz audio, so no, not more that 128ms, but definitely more than 16ms. No, 128ms = 1024 taps Like what sangoma offers. Ding, Ding, Ding, Ding! ___ --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] Echo Problem with T411P
Steve Davies wrote: On 6/15/06, Idris AVCI [EMAIL PROTECTED] wrote: Hello, There are 3 PRI's connected to the card each from different operators. Especially echo occured on span 3 is really annoying. Configuration files are as follows. Is there something wrong in conf ? Have you verified that the provider on span 3 is not using some high latency routing? The configuration line context=Satelco suggests a satellite company? They should do the echo cancelling on your behalf if they have high latency routes as the asterisk EC will never keep up under those circumstances. We have even experienced problems within Europe where providers route national calls via international routes to save money. This adds significant latency and makes any echo so heavily delayed that asterisk cannot remove it. More than 128ms? ___ --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] GXP-2000 addressbook
Matthias Fechner wrote: Hi Gareth, Gareth Blades wrote: No I dont believe so. The address book is a new feature as it is very basic in my opinion and even editing it on the phone is difficult. I would expect a web based editing feature to be implemented at some point and once that is done it should be possible to do a mass update of the phones. ah ok, then I will wait for a new firmware :) This is one of those times where you should be contacting the supplier you bought the phones from. They should be able to get your message over to grandstream so they know what people want. Other than better phones of course. ;) ___ --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] EC needed in all-digital situation?
Eric ManxPower Wieling wrote: Warren wrote: I was just told that for my forthcoming system I will be getting a data T-1 instead of a voice T-1. Given that all of the handsets will be voip phones, no analog at all, do I need echo cancellation? I looked at the voip-info wiki and it seems to me that the answer should be no but I would like to confirm that. If you never make calls to analog phones or receive calls from analog phones, then you will never have echo. Can you be sure that all telephones you call out there on the world PSTN will always be digital? I didn't think so. If you have no TDM equipment, then it is your provider's responsibility to handle any echo that may be generated if a call is routed through the PSTN. There is more than one way for echo to naturally occur that have nothing to do with TDM circuits, like bad acoustics in a room as mentioned by another poster. VoIP moves much of the quality needs from the center of the network to the edges. That is, phones and gateways. ___ --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] Easiest (best?) linux distribution for dedicatedAsterisk box?
Tzafrir Cohen wrote: On Wed, Jun 14, 2006 at 01:15:03PM -0700, shadowym wrote: FreePBX or AAH(aka trixbox) requires 256MB of RAM minimum to run properly. Just sitting there doing nothing on my test system it is using 170MB. How exactly do you meassure memory usage? E.g: on my laptop: $ free total used free sharedbuffers cached Mem:483056 476320 6736 070820 169360 -/+ buffers/cache: 236140 246916 Swap: 976744 3048 973696 Technically you could say that it only has 6.7 MB free. But actually some 220MB are used for buffers and caching by the kernel (because unused memory is wasted memory). BTW: where did you get the idea that Asterisk == FrePBX? Asterisk is not known to require MySQL and Apace to run. My home debian linux box runs, iptables, asterisk, apache2, exim, dovecot, dhcpd, proftpd and tftpd on a PPRO 200Mhz with 64MB ram for two users. Yep, it swaps a bit, (especially when running something big like emacs), but it does what it needs to do and it doesn't have a high load on it. The only thing I'm not running is mysql and astmanproxy. I doubt centos requires much more memory than debian. ___ --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] Comfort noise support incomplete in Asterisk (RFC 3389)
Comfort noise is the sound you hear from the phone to assure the user that there is still a connection to the other end. It is there to keep you from hearing no sound through the speaker and thinking you have been disconnected. Check your phone's config for comfort noise or silence suppression and turn it on or off respectively. What phone model(s) do you see this with? Daniel Salama wrote: Can anyone explain to me what this means: Jun 14 19:46:10 NOTICE[7391]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 66.175.1.1 When I try to make a call from certain IP phones, I see that message on the console. Thanks, Daniel ___ --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] SPA941 and Echo
Try reducing the gain on the microphone. These phones pick up room sounds *very* well. Andres wrote: Has anybody else experienced bad echo issues with this SPA941 phone when calling SIP-SIP to another SPA ATA? When I call remote office phones that are attached to SPA ATAs, I get very annoying echo. One can sure blame it on the reflected signal from the phone on the remote end, but how can one deal with this echo? (it happens with some phones really bad and with others its perfect) The SPA941 does not handle echo itself so I am at a loss on how to approach this issue. If I change the SPA941 for a regular SPA 1000 with Echo Cancel enabled, I can hear perfectly. Any ideas? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA941 and Echo
Andres wrote: Mike Fedyk wrote: Try reducing the gain on the microphone. These phones pick up room sounds *very* well. WellI'm not using the speakerphone. Plus there is no gain setting at all that I am aware off. Just Handset Volume or Speaker Volume. I'm not talking about the speakerphone. Check the XML config file. ___ --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] delay in MeetMe
Get some hardware, a TDM410b is only $125. Or upgrade to 2.6.13 or later. Don't compile the kernel unless you know what you are doing. You might try, Ubuntu 6.06, FC4 with updates or FC5 to see if that makes a difference. Also there are patches on mantis for delays in meetme conferences that are unrelated to timing. See if -q makes any difference, and if it does, those patches should help. amna saleem wrote: Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem ___ --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] open source sip softphone (Window OS version )
Asterisk guy wrote: are there any open source sip softphone (Window OS version )? http://www.voip-info.org/wiki-Open+Source+VOIP+Software ___ --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] Web UI - Best practices?
Tzafrir Cohen wrote: On Wed, Jun 14, 2006 at 11:51:06AM -0400, Mike wrote: Hi, I'm stuck writing a Web GUI because nothing out there is exactly what I need. I'm not writing something as extensive as what _is_ out there, but just something that allows users to change where their calls are forwarded and other small things like that. What I wanted to know is what is recommended by those you successfully wrote their own UI : 1) Modifying the config directly in the Asterisk RealTime DB and and use Asterisk Realtime? This seems like the obvious choice, but I have a bad feeling about this method...especially with respect to future changes I would make to my UI or that the Asterisk dev team would make to their own tables / code Non-static real time mean that your PBX becomes non-functional if the DB server has a problem (or is even a bit loaded). Why is it that more people haven't turned to LDAP to solve this problem? It is optimized for a light write and heavy read workload that seems perfectly suited for use in a PBX dial plan. ___ --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] Can this config sustain 30 users?
Erick Perez wrote: I just don't want to install it and then after a 5th user going to call someone the asterisk begin to crash due to lack of resuources. Check the wiki for SIP load generation tools you can use to test your setup on any number of calls you like. ___ --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] Can this config sustain 30 users?
Martin Joseph wrote: Ultimately you need to set up a server that does what you need and see how it performs. Usually hardware overkill is a good bet, but you don't need to go crazy. So, one cpu per call is too much? ___ --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] Can this config sustain 30 users?
Erick Perez wrote: I have this server I need to put to work. The option I have is to make it work as a small office PBX with SIP users and a Digium E1 Card for PSTN service. 24 SIP users and one E1 card in an Intel 945board (533 Front side bus) with 1GB DDR 533mhz of ram, one Pentium Dual Core 2.66 ghz (FSB 533MHZ) and two 80GB SATA disks. Can the box sustain the load? I can add another 1gb of ram if necessary. Just PBX and voicemail, no fancy sutff like call recording... maybe a simple autoattendant like thank you for calling, please press one for Let's just say that you shouldn't have trouble with four E1 lines in this system (as long as you have a hardware echo canceler). Even with software echo can, this hardware is overkill. I'd suggest getting the slowest processor available new (At least 1.5Ghz for AMD Athlon/Opteron and 2.xGhz for Intel P4/Xeon) and get redundant power supplies, ECC memory, RAID 1 or higher and a nice UPS. You may also think of looking at used previous generation high-end equipment. It still has all of the good engineering and redundancy, it just isn't new and in many cases you don't need the latest speedy equipment to handle the load. In many instances, the same money spent buying used high quality equipment will give you more reliability than buying cheap new equipment. ___ --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] Snom high SIP ping time
Steve Glaus wrote: Mike Hammett wrote: (ICMP) pings were under 1 ms. No amount of different Asterisk versions or phone firmware revisions seems to solve this. All was well, then (as far as we know) without changes, it crapped out. Any ideas? I'm having much the same issues only I'm using Cisco 7960 phones. When I do a 'sip show peers' I'm getting times in excess of 300ms. A soft phone on the same network (x-lite), is reporting times of 4 ms. Related to this (I think), I'm getting audio issues. The person being called can hear the caller fine but the callee's voice drops in and out excessively. I have qualify set to yes in the sip definitions for all the clients (Including the soft phone). Does anyone know what is causing this. I'm not aware what the sip ping times were earlier, but the audio issues seemed to have started spontaneously. Do you have any problems when there are a low number of concurrent calls? Do you ever get any messages saying the phones are unreachable? Or lagged? What kind of Internet connection do you have? Do you have any problems with calls between phones on the same network (no routers in between)? What model of switches do you have? What model of Internet router do you have? ___ --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] Are zttest results relevant on a system with no telephony hardware?
Carl Youngblood wrote: Our asterisk system gains access to the PSTN through a voip provider. We have no digium or other telephony hardware in our system. Do the zttest results still matter to us? Our results were as follows: --- Results after 1007 passes --- Best: 100.00 -- Worst: 99.780273 -- Average: 99.975763 IAX trunking and meetme conferences are some of the heaviest users of zaptel timing. I'd suggest if you don't have hardware timing (or at least a 2.6.13 based kernel), then use SIP all the way or at least turn off IAX trunking. ___ --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] Can this config sustain 30 users?
http://www.voip-info.org/wiki/view/Asterisk+dimensioning http://www.voip-info.org/wiki-Open+Source+VOIP+Software Erick Perez wrote: I appreciate all your help and posting. I will then load (with test calls) using SIPP and astertest will post back the result of this machine in question any other open source stress test tool i can use? Thanks, On 6/13/06, Colin Anderson [EMAIL PROTECTED] wrote: I'd suggest getting the slowest processor available new (At least 1.5Ghz for AMD Athlon/Opteron and 2.xGhz for Intel P4/Xeon) I'm fond of underclocking. No heat problems for me, thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are zttest results relevant on a system with no telephony hardware?
2.6.13 has the high precision timers changes. And people have reported 99.99 averages with that change. I do not recommend compiling your own kernel unless you know what you are doing though. If you just want to give it a try, then use Ubuntu 6.06, Debian Etch, FC4 or FC5 (or some other distro that packages 2.6.13 or higher). If you want to use it in long term production, then just get a TDM410B card and use hardware timers. Carl Youngblood wrote: Thanks. What is it in the 2.6.13-based kernel that improves timing? Should I expect to see a significant improvement if I upgrade to it? On 6/13/06, Mike Fedyk [EMAIL PROTECTED] wrote: IAX trunking and meetme conferences are some of the heaviest users of zaptel timing. I'd suggest if you don't have hardware timing (or at least a 2.6.13 based kernel), then use SIP all the way or at least turn off IAX trunking. ___ --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] Linksys SRW224P POE Switch
Tom wrote: Most of the latest generation POE switches including the Linksys SRW224P provide their power on the data pairs, not the unused pairs. So if both the data and the power are on the same pairs, how do you make a cable adapter to work with the 7960G? Maybe bridge the unused pairs with the data pairs? I haven't tried it as I don't have any old style PoE, but it seems plausible. ___ --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] Easiest (best?) linux distribution for dedicated Asterisk box?
First, remove telnet from your vocabulary. It should only be used over serial connections these days. All other times, you should be using ssh. Second, do you want the computer to be installed and running without any major software changes for a year or more? Then use Centos or Ubuntu Dapper 6.06 or Debian Sarge 3.1. Make sure you don't install the graphics as it can affect the latency of asterisk, especially on older hardware. Third, I run asterisk on a PPro 200 at home, so your machine is beefy enough for sure. And lastly, just give it a try, you'll learn a lot just making the effort. Mike John Klimek wrote: First off, I'm sorry for sending so many messages to the list-serv. Hopefully this will be my last for a while! I was going to use my WRT54G router as a small Asterisk box, but I forgot that I had a spare eMachines computer (Intel Celeron 633 MHz, 20GB HD, 64mb RAM). Will this machine work OK for a very simple dedicated home Asterisk box? Also, what is easiest linux distribution to use and install? All I want is a simple Asterisk box that I can telnet into and have voicemail, music-on-hold (MP3), etc... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] IAX2 Vs SIP cpu load
Patrick wrote: On Tue, 2006-06-13 at 23:47 +0800, Dinesh Nair wrote: On 06/13/06 22:49 Colin Anderson said the following: Although this may have changed in the newer 1.2.X series of Asterisk, I believe that Asterisk does not support SMP from the perspective of isnt asterisk multithreaded ? on a proper OS thread implementation, threads can migrate across CPUs, can't they ? Afaik in 1.2.x IAX is single threaded. In 1.4 it is multithreaded. In 1.2.x IAX uses two threads. One to send, and one to receive. In 1.4 it will use more threads, but I don't know what the new threading model for IAX will be though. ___ --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] spa3102 vs spa3000 differences?
Steve Davies wrote: On 6/12/06, Doug Crompton [EMAIL PROTECTED] wrote: It seems that any firmware is usable on any hardware as my hardware is 2.x. I wonder if 3102 firmware could be used on the 3000. Is the size the same? I guess you would have to be willing to make a brick to find out! I have not tried this, but on an spa2000, the firmware updater simply made no changes when I tried to install some unsupported firmware. Tim told me that if you change the DC adapter for one with a couple more amps the firmware flashes faster. More Power! ;) ___ --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] Hard drive write cache
shadowym wrote: Any other recommendations/links for increasing the reliability of Asterisk servers? Separate the various use cases of the filesystem into different volumes with LVM. The parts that are not written to except during upgrades like /usr should be mounted read-only, and the various read/write sections like /var/spool/asterisk and /var/log should be on separate volumes also. This keeps any corruption experienced localized to a small area, and keeps your binaries unquestionably safe from a power outage and the only step needed is automated detection and cleanup of the read/write volumes. Write barriers allow you to keep write-cache turned on in the drives, and sends a command to the drive to reply when the data hits the platters. Also if the drive doesn't support that, various techniques are used to verify the data is on the media and not in drive cache. Contact your distro support company and ask them if write-barriers have been implemented in the drivers for the drive controllers you are using, and if not, then either buy the hardware that does support that or sponsor them to update the driver to support write-barriers. Also contact your distro support company to see if they have any recommendations for the setup you want. Mike ___ --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] Reorganizing menus in Polycom 301? Was: [asterisk-biz] New Polycom SoundPoint Series IP-430
Chris Mason (Lists) wrote: Cory Andrews wrote: IP430, will sit between the IP301 and IP501, IP430 will have dual Ethernet, PoE, and full duplex speakerphone. List price (MSRP) $239 street price should fall likely between IP301 and IP501. That looks great, the 301 is almost useless due to the lack of speaker phone and message buttons. Agreed and the menu organization sucks. Anyone know if there are any hacks to change the menus in the IP301 phones? I'm specifically looking for one or two button access to the redial list (reprogramming the redial to display dialed list and then have second press redial last number would be great). Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO registration and VegaStream
Peter Doyle wrote:
I figured asterisk was looking for SIP user 06, so I added it, but I
still got 404's. Turns out I just needed an EXTENSION, 06. I can now
make calls and receive them, too. Of course, if you have multiple
incoming lines, you'd need extension 06, 07, 08 ... etc, since each port
has its own Interface Number (by default), to allow routing of calls
made to different lines.
Yes, that's right.
You should specify a separate context for the incoming lines and if your
port numbers relate to extensions do something like this:
exten = s,1,dial(sip/20${EXTEN},,o)
That way when you dial port 01 on the vegastream, it will ring on the
sip extension 2001.
This page explains the s context more:
http://www.voip-info.org/wiki/index.php?page=Asterisk+s+extension
___
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Re: [Asterisk-Users] Fun with Echo
Steve Davies wrote: On 6/9/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote: Consider getting a Sangoma A200D (http://www.sangoma.com/datasheets/p_a200-specs) with the optional hardware echo canceller module. It just works for echo cancellation; no tweaks required. It takes a while to figure out how to install it, but once it's working it's great! Yes, I must agree that hardware echo cancellation from Sangoma (and I am sure Digium and others) is excellent, but it does add to the cost quite significantly sometimes. If you haven't had your client sign a waiver that there maybe echo because of the cheap hardware, then don't use the tdm400p. I've wondered why the software echo can in zaptel doesn't go up to 1024 taps and causes major problems at 256 taps. The tdm400p cards don't have a high port density, you'd run out of slots before hitting CPU barriers so why not use some of those resources for an echo can with a longer tail? Use either the sangoma cards with echo can, or get a tdm2400p with echo can. You will have happier customers if you do. ___ --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] Quad T1 Card
Kevin P. Fleming wrote: - Matt Riddell (IT) [EMAIL PROTECTED] wrote: What does the onboard DSP do when used with Asterisk? Did Digium or someone put code inside Asterisk to hand off the processing/transcoding to a Sangoma card? According the Sangoma data sheet, the Octasic part _is_ the DSP (which it is, in a logical sense). The board does not relieve Asterisk/Zaptel of any additional burden beyond echo cancellation and tone detection at this time; Asterisk/Zaptel don't know how to take advantage of any of the more advanced Octasic features yet. And yes, when Digium's Octasic-based module starts shipping (currently in beta testing), it will offer the identical functionality, so I guess we can say our boards have 'DSP processing' too :-) Will it have a 1024 tap echo can on all 96 channels? What about 8 T1 support like sangoma? ___ --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] Quad T1 Card
Kevin P. Fleming wrote: - Mike Fedyk [EMAIL PROTECTED] wrote: Will it have a 1024 tap echo can on all 96 channels? What about 8 T1 support like sangoma? Those are completely unrelated questions; there is no need for an 8-span echo can module when there is no 8-span T1 card :-) It uses the identical Octasic part as the Sangoma board does, so the capabilities will be the same in that regard. Have you seen the A108? http://www.sangoma.com/press/corporate/2006_04_05_A108_Card ___ --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] a new asterisk version
http://www.asterisk.org/download http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+CentOS amna saleem wrote: Hi All, I need a suggestion. I want to run only IAX on two windows based PCs and asterisk Can you suggest which asterisk , libpri and zaptel versions should i use? do i need some other modules also? Also where will i find the guide to compile astreisk Actually i have installed,comnpiled and used astreisk-1.0.3 on Red hat 9 which was not that stable. Now i have Red hat Enterprise on my PC. i think there are newer stable versions which can run on Redhat Enterprise Linux. Kindly help, ___ --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] GXP-2000
I have a client who has about six of these phones. Luckily (for me, not for them) they were purchased before I came into the picture. Daniel Salama wrote: I have heard complaints from my client about the speakerphone and they are now You don't notice any problems when using the speaker-phone, but the person on the other end hears echo, and quite a lot of it. , I guess, getting used to picking up the handset :). My client uses them exclusively with headsets (in a call center) so the quality of the speaker-phone isn't an issue for them. I have heard any echo problems so far. What bothers me the most is that the phone stops working often (multiple times per day). By this I mean that my client won't be able to dial anything successfully. As soon as 3 or 4 digits are entered, they get a fast busy. To solve it, they need to reboot it. It sounds as if these phones were running Windows instead of Linux :) Do you have multiple phones going down at the same time? If so, monitor them with qualify=500 in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. Also I have heard some phones have trouble with broadcast packets (at least this has been said about the spa-841 on the wiki). You should strongly consider putting them on a separate vlan to avoid any issues like that. In the future, for phones under $100 then look at the spa-841 phones. Anyway, what firmware did you use that solved so many of your problems? http://www.voip-info.org/wiki/view/GXP-2000 ___ --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] GXP-2000
What specifically were the voice quality complaints about the spa-841 phones? The only thing I have noticed is calls can be louder than expected. What else have you seen? Daniel Salama wrote: They don't all go down at the same time, or at least, my client hasn't noticed. I just added the qualify option. Let's see how that goes. As for the SPA-841, I have a client with a few of them and he cannot stop complaining about the bad audio quality. I replace a couple with a PAP-2 and another one with the GXP-2000 and he claims the quality to be incredibly better for both the PAP2 and the GXP-2000. He hasn't complained about the problems I mentioned on the GXP-2000 - yet :) Thanks, Daniel On Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote: Do you have multiple phones going down at the same time? If so, monitor them with qualify=500 in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. Also I have heard some phones have trouble with broadcast packets (at least this has been said about the spa-841 on the wiki). You should strongly consider putting them on a separate vlan to avoid any issues like that. In the future, for phones under $100 then look at the spa-841 phones. ___ --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] GXP-2000
Did you try setting the RTP packet time size to 0.020? Also I would look at the trunk, provider or internet connection before the phones I started suspecting the phones. I have had the same problems with providers, and the conversations sound great from one location to another over the internet, but once it hits a provider, the sound quality drops. That is not the fault of the phones. Are you sure you didn't change anything else when you switched from the spa-841 phones? Daniel Salama wrote: The complete opposite. The user complaints that either they cannot hear the remote party well or the remote party cannot hear them well. Sometimes it works and sometimes the volume is very low and that's why they cannot hear. - Daniel On Jun 7, 2006, at 1:35 PM, Mike Fedyk wrote: What specifically were the voice quality complaints about the spa-841 phones? The only thing I have noticed is calls can be louder than expected. What else have you seen? ___ --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] Good ATAs from companies other than Sipura/Linksys?
First of all, I'm not knocking Sipura/Linksys. I have heard very good things about their products. I'm just wondering if they are the only quality shop on the market. I know about the zoom 5801 where you can't dial out the FXO from SIP, only from the FXS port. And I have heard similar about the HT-488 also. I want to know if anyone else makes ATAs where all of the features work as advertised. If it has two FXS ports, I want to be able to use both at once, from Asterisk, same for T.38 and etc. So, what's out there that I don't know about in the world of quality ATAs? ___ --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] GXP-2000
John Novack wrote: Is the 94x any better? seems without backlighting, any are next to useless. Yes, I like the 941 better than the Polycom 301 and the display is much improved (no backlight, but one of the guys at voipsupply told me that the 942 has a backlight which sounds very promising). The base for the 941 is more angled like the polycom phones and it is bigger and heavier so it doesn't move around as much. And the buttons have a very nice feel. With the list of phones I have used, here is how I would choose them (first being better): Polycom 501 Linksys spa-941 Polycom 301 Sipura/Linksys spa-841 Grandstream GXP-2000 ___ --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] GXP-2000
Kerry Garrison wrote: I would never ever ever sell a client on a SPA-841 or heaven forbid the GXP-2000. All the clients who bought those originally sold them off and went for better phones very quickly. Let me say that when suggesting the spa-841 it is only in the context of sub-$100 phones. I hadn't worked with any spa-841s before, but when my client wanted cheaper phones than the 941s that I suggested, I strongly warned them that from what I had seen, about 50% of them are returned. But they insisted and I have to say that the phones are not *that* bad. There a lot of things I like about them that I don't like about my polycom 301 (though most of my gripes with the 301 could be fixed by remapping some of the buttons and make call lists available with one button press, so it's not a hardware deficiency except for the lack of speakerphone and backlight). Mike ___ --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] GXP-2000
I have heard good things about the D-Link DES-1226G switch ($150 at newegg). If you can run a separate cable to the computer and phone. If you can't run the extra cables, then configure your phone to tag itself as part of the voip vlan and let the switch tag everything else as the computer vlan. I happen to have asterisk running as a router, so I use it doing QoS with tc (traffic control) and wondershaper set to prioritize based on port ranges. I sent a patch to the debian bug tracking system a while back with a few improvements -- I should check on that. It basically prioritizes smaller packets before larger packets with ~8 levels of priority and groups of sizes for the packets. Just doing that automatically handles 80% of the need for prioritization without specifying port ranges for the sip rtp packets. Mike Daniel Salama wrote: They are extremely casual web surfers. Just have their Outlook client opened checking email every minute. Email traffic is very low. They are all connected to the same switch. It's a Netopia DSL router/modem/switch for the BellSouth DSL service. The computers are connected to the PC port behind the GXP-2000. Any suggestions? Thanks, Daniel On Jun 7, 2006, at 8:49 PM, list mail wrote: What do they do on the internet? Heavy surfing, large transfers, myspace. How are these units connected to the network? Are they passing through the same switch? I don't think it is the phones... On Jun 7, 2006, at 12:32 PM, Daniel Salama wrote: Mike, I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM: Jun 7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms) Jun 7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (66ms / 500ms) Jun 7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms) Jun 7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (68ms / 500ms) Jun 7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now TOO LAGGED! (1114ms / 500ms) Jun 7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now REACHABLE! (90ms / 500ms) Jun 7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1077ms / 500ms) Jun 7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (72ms / 500ms) Jun 7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now TOO LAGGED! (1077ms / 500ms) Jun 7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now REACHABLE! (73ms / 500ms) As you can see, it only happens to a couple of their phones and at random times. They're behind a DSL circuit. I don't know if it's because their DSL line is going up/down. They don't necessarily claim their Internet goes down, however, they are not constantly check it. What would you (or anyone else) suggest? Thanks, Daniel On Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote: Do you have multiple phones going down at the same time? If so, monitor them with qualify=500 in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. ___ --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] Prices of g729 codec
Kevin P. Fleming wrote: - Jon Lewis [EMAIL PROTECTED] wrote: IMO, locking the licensing to a piece of system thats often built-in, has been very annoying. I think I'd be happier if it was locked to some sort of dongle (parallel, or more likely today, USB). At least that way, we could easily move the key anytime we needed to. It would be a bit of a pain any time a system needed to quickly be transfered to hardware already at another location. I have proposed that a number of times internally, only to be told (vehemently) that customers would never go for it. That includes responses from our distributors and channel partners, among others. It would also dramatically increase the cost for people buying one or two licenses, so it would have be an 'alternate' registration means if it existed. How hard is it to use a removable ethernet card for this type of usage? Also a USB ethernet if with Linux drivers should be usable for the 1U rackmount use case where all internal slots are in use. Most of the complaints should be able to be remedied by and update in docs for recommended implementation (removable ethernet, either PCI or USB). And just make sure the g729 codec can see those ethernet ports. ___ --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] Looking for postpaid quality A-Z termination
In other words, please post your message to asterisk-biz instead. Martin Joseph wrote: What part of NON-COMMERCIAL do you not understand? ___ --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
Adding Asterisk between existing phone system and PSTN Re: [Asterisk-Users] Integrating Asterisk
Dakota Burns wrote: What I was attempting to visualize is the following case: 10 people in an organization pick-up their phones to make an outbound call. Before integrating Asterisk, all calls route through their current non-VoIP based phone provider. After integrating 1 trunk from a VoIP service provider into their system that provides 4 simultaneous calls (Teliax's Corporate plan) First of all, don't use a service that limits the number of calls. Get the per minute plan. That way you won't have to worry about hitting any soft-caps. , and dropping 4 legacy lines Don't drop the lines until you have a setup that works reliably without hickups for at least 3 months (more if you can convince them to keep the lines that long). , if 10 people make calls simultaneously, some will be VoIP and some will be legacy based. Based on the above example, I'm questioning whether it would be best to configure a Sipura 3000 for every analog phone (I'm guessing the non-profits will want to keep their existing analog phones) Only use ATAs when you have to. They cost about $80 anyway, why not get a spa-841 instead? And why are you guessing? You should know if they want to keep the phones they have. And what type of phones are they? , or utilize another device (or devices) to connect the company's internet service into their existing Trunks or POTS. I think the former would be easier something I know how to do, but the latter may be smarter more cost effective. So the latter is what I'm questioning whether either of you have experience implementing. Let me be frank. I'm relatively new to phone systems, but I can tell you need to do a lot more research before even thinking about doing an implementation. If you want to keep the analog phones, they probably already go to a wiring closet. You'll want to put either an asterisk box with a tdm2400p with 12 FXS and 12 FXO (look up the tdm2400p before asking why I say 12 instead of 10). Or if you have voice T1s at that location you may want a channel bank instead. I haven't used any channel banks so others will have to step in to give suggestions on that. My point is that you need to post what you want your client's results to be instead of how to do what you think should be done. The details I mentioned above are only part of one possible direction to go in, and there is more to it than that also and it may not even be the best for your situation too. Have you looked at their network to see if can handle the large number of small packets that voip produces? What about their Internet connection? What is it that your client wants in a phone system that their current one isn't doing? How is adding asterisk and an ATA for each analog phone going to help? So, post what you already have and what you want the end result to be from an end-user's perspective and we can probably point you in the right direction. ___ --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-Asterisk hints/presence
How do you setup asterisk so that the assistant sees the lights but
doesn't hear the rings?
picciuX wrote:
damon, i think many guys here missed your point and went away from it.
What you want to do is possible: i managed to do that using a GXP-2000
with beta firmare and asterisk 1.2.0.
GXP correctly processes the status change messages and show a
blinking led for the BLF of a ringing extension. The GXP, when you
press a blinking BLF, dials out the blf extension prefixed with
'**'. In the dialplan i only needed to do a:
exten = _**XXX,1,Pickup(${EXTEN:2})
to answer the ringing remote extension.
So, IMHO, it's only a matter of the phone. This is not a true shared
line, but does its job well letting an assistant answer his boss' phone.
AFAIK, Snoms do a similar thing, dialing out *8BLF extension when
pressing a ringing button.
Hope this helps
2006/6/2, Damon Estep [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
I think there was a patch that went in recently from Mark with
regard
to SIP devices and their state when they are ringing/in use and when
they are just in use. That may help you with what you're asking
about.
Let's assume for a minute that there is a way to get a ringing notify,
and that the Polycom processes the messages and properly displays the
status, then;
1. A hint could be setup that monitors the sip user.
2. the hint could have a unique extension
3. the dialplan for the extension could be a call pickup sequence.
Example (without regard for correct syntax)
Exten 123,hint,SIP/345
Exten,123,1,(Need help here - if sip/345 is idle then goto
priority 3)
Exten 123,2,Pickup(SIP/[EMAIL PROTECTED])
Exten 123,3,Dial(SIP/345)
Now, on the Polycom
1. setup a buddy with the correct display name for exten 345, but
with,
but with a contact address of 123
2. turn on presence monitoring (buddy watch in Polycom terms)
Do you see the value in knowing that an extensions is ringing via
presence?
Take it one step further, ask Polycom to implement a feature where the
presence status of ringing can produce an audible event instead of
only
visual events. That is, if it does not already, which I have no
idea on.
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Re: [Asterisk-Users] Polycom-Asterisk hints/presence
Damon Estep wrote: I understand your technology agnostic position, and it makes sense, however my vote (for the little that it is worth) would be to implement a SIP rfc complaint shared line appearance capability (and/or bridged line appearance), and then, if possible, extend it to support zaptel and iax and whatever else is popular. SIP is arguably the most common choice for NEW VoIP implementation, and it also appears to be the common ground upon which all vendors of VoIP gear will meet for interoperability. IAX, even with its advantages, will not likely progress to the same stage of universal acceptance, it may very well be the choice of many asterisk users, but in the end you will still have to talk SIP interoperate with the VoIP Revolution Ok, let me jump in and explain it a different way. The way asterisk works is it abstracts concepts from protocol details. For instance let's say a protocol like SIP or IAX is a human language (and what is a protocol except for a means of communication between computers like language is a means of communication between humans?). They both have concepts for the concept of walk but the English and Chinese languages implement it very differently. I think the part that most people are missing is being able to monitor the status of another extension (not channel, since channels are only in use during calls) and ringing multiple phones simultaneously can easily be done with queues or dialing multiple extensions from the dial() command. Now let's say all languages (protocols) have the base functionality of lighting a call appearance when it is in use, and blinking it when it is ringing. Then an assistant can know when their boss is on the phone or not at a glance. These base concepts need to be implemented in the core so that they can be interpreted by the protocols (languages) into the specifics of what hits the wire. So, that is why it must go into core and then that needs to be exposed to the protocols. The hardest part is finding the right abstraction so the concept (core) can be translated properly into the various languages (protocols). And not one reference to a car or driving. ;) How's that? Mike ___ --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] addons trunk make error
There are too many changes happening in trunk to constantly update -addons to work with it. Once things settle down a bit, they will bring -addons up to date. This has been repeated a few times in asterisk-dev recently. Did you google for asterisk trunk addons compile? Damon Estep wrote: Anyone run a make on asterisk-addons /trunk r219 ? I error out on mp3 on a FC4 box, and I do not see anything obvious (to me) in the errors. ___ --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] SMP kernel on Pent 4?
Rich Adamson wrote: Had a Pent 4 server running fc3 crash (kernel panic) and am rebuilding from scratch. I installed FreePBX (CentOs) from scratch and asterisk was running, but had not yet been configured. It too crashed with a kernel panic. Ran memtest for 24 hours; no errors or issues uncovered. I then noticed that FreePBX installed using a SMP kernel (and grub indicated a non-SMP kernel was installed as well). Would running an SMP kernel on a Pent 4 potentially cause a kernel panic? (Or, do I need to dig somewhere else?) Nothing in the logs to suggest a root cause and I'm now waiting on recurrence using the non-SMP kernel. Were you able to see an oops message when it crashed? If not, then make sure a X11 server isn't running, and turn on nmi_watchdog. The easiest way to capture the oops is with a serial console, but hand typing the text into another computer or a snapshot has worked in the past also. Then post your results. Also check the system temp with lm_sensors and the quality of your drives with smartctl. Mike ___ --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] Two asterisk process in one hardware.
Juan Salas wrote: Hello. Has anybody knows how run two asterisk process in one hardware? (each one with its own configuration?) What end outcome do you want? Maybe there is another way to do 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] Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)
I have a client with an installation with 3 TDM400P cards. 6 FXO, 6FXS ports. I followed the txgain/rxgain instructions and now have no echo problems. The only problem I have now is the flaky network the SIP phones are accessing asterisk with. (you should see the wiring there, ugh). It's in a dell p4 desktop system. I don't recall the model, but I can find out when I give them a visit next (or if you'd like lspci output I can do that now..) Mike Jared Davison wrote: I would like to hear from anyone good or bad as what their experience has been in recent times with STABILITY of current builds of Asterisk and drivers for TDM400P. The sort of configuration is: 6 incoming POTS lines. ie. 2 TDM400P cards. I am not concerned with: price points, or the advantages or disadvantages of using POTS vs ISDN technology, but simply RELIABILITY stability of the Asterisk system associated interface hardware and drivers. Do people need to reboot their systems regularly? Thanks in advance. Jared ___ --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] Re: Mediatrix windows-based setup?
No, you replied to a message from Vladimir Montealegre with the subject Re: [Asterisk-Users] RJ21-RJ11. That is called thread hijacking. You may sort your mail by date, but others use a feature called threading. It keeps track of who replied to what message to be able to see a conversation in several email messages. If you reply to a message instead of creating a new message, it really messes up the threading. Please don't do that. Kerry Garrison wrote: What are you talking about? That is the address I used. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Parcina Sent: Monday, January 16, 2006 6:22 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Mediatrix windows-based setup? Please stop replaying to mesage. If you plan to open thread do so by writing mail to this address asterisk-users@lists.digium.com -- Tomislav Parcina [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 ___ --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] Re: Failover Device?
Matt wrote: On 1/12/06, Tomislav Parcina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... First, Something seems to be wrong with the list. I'm not the only person who has expressed seeing their messages either arrive late, or not at all. I'm sure that I'm not the only person that has notice that there is lots of people that start new thread by replaying to old message. That way neither them, or lots of other people, sees that mail as new therad. Yeah I've noticed that too.. I don't do that though. Ok on to the question at hand. I am trying to fail over asterisk. I have PRI redundancy. What I need, however, is someway to transfer the PRI from asterisk box A to asterisk box B if asterisk box A fails. So while, yes, I can build a second asterisk box and use SER, or DNS or whatever to point my sip devices to it... the question is how do I get the PRIs to know which box to route to? Check out red fone. It is designed to work with Asterisk in a HA configuration. Though, if you only have one, it will become the single point of failure. It's about the same price as a TE411P with echo cancellation. http://www.red-fone.com/fonebridge.html You have two PRIs already, that can be used for fail-over. Put a TE110P in each server (or TE411P if you need echo cancellation in the card or plan on expanding) in each server. If one fails you have your provider send calls to the other PRI if one fails. ___ --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: Nested MySQL Commands
Connection pooling doesn't require threading. You can also use a pool of processes which are quite cheap on Linux. Douglas Garstang wrote: Do you have a link to where it says this? The DBI docs that I looked at (perldoc dbi) said that it isn't thread-safe. -Original Message- From: Leo Ann Boon [mailto:[EMAIL PROTECTED] Sent: Thursday, January 12, 2006 12:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Nested MySQL Commands Douglas Garstang wrote: I also don't believe perl DBI is thread safe The lastest docs says that DBI does support multithread connection pooling. Otherwise, you are always free to implement your AGI in 'modern' :) programming languages like Java or C# that support threads and pooling. -Original Message- From: Douglas Garstang Sent: Wed 1/11/2006 9:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Re: Nested MySQL Commands Since about 1992... and the Asterisk docs for FastAGI are pretty rotten. But that's ok, I've come to expect that. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Wed 1/11/2006 8:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Re: Nested MySQL Commands Douglas Garstang wrote: I don't get the whole concept of FastAGI. It's nothing special. Asterisk just opens a connection to a TCP port instead of executing a binary. How long have you been around Unix/Linux systems? Do you have any clue how much less expensive it is to open a TCP socket as compared to forking the Asterisk process, exec()-ing another program, having that program open database/web connections, etc.? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Nested MySQL Commands
Simone Cittadini wrote: Douglas Garstang ha scritto: So I really wish there was some way to measure how well the worst case scenario would perform. This would be 120 simultaneous calls (don't know how many per second) on a Dual 3.8Ghz Dell PowerEdge 1850 with 2GB RAM. Asterisk would call an AGI script, written in perl, to route all calls. The script would have to perform multiple database queries in order to route a call. It will work if you need no transcoding, I tested a python agi doing something like 6 query to accept / instradate the call and it works for 150 / 200 simultaneous calls, the machine starts sweating of course, but the voice quality is still good, no drops. Mine is just a quick prototype, using fastagi or writing the agi in C is surely the way to go, imho fastagi will let you have a more configurable / customizable system since you can write the application in a object oriented language. Also an ugly hack would be to call the perl bytecode instead of the text script. That would allow for the ease of AGI (everything is cleaned up when the process exits) with lower overhead. FastAGI is of course what you want for production, but this can help in a pinch. Mike ___ --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] Nested MySQL Commands
Andreas Sikkema wrote: Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. Just make sure that you use different variable names for each query if the values should stay available after the next query. What we tend to do is grab the data from the database and the stuff that should stay around for a longer time is assigned to a new and appropriately named variable. So the original variable can be used again. I'd rather use a hash (also called associative arrays in other languages) instead of calling eval to assign a variable. ___ --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] Nested MySQL Commands
Chris Albertson wrote: Under Linux (and other OSes) It's not as bad as that. Even with 128 Perl processes running there is only one copy of the Perl interpeter in memory. Each of the 128 running processes would have it's own copy of only it's data segments. With Perl already in memory the biggest system overhead would be process creation. The best design is the one that minimizes the number of process that the kernel has to create. Notice that this is why the Apache Perl modual is so much faster than using Perl from a CGI script You will get the best usage of shared pages if all child interpreted processes fork off of one parent process. That way they can share as many data pages as possible also. If they don't fork off of each other, then a new copy of the interpreter will be put into memory. There will be some shared CoW pages between them, but not nearly as many when compared forking off of a common pool of processes. ___ --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: Nested MySQL Commands
Tony Mountifield wrote: In article [EMAIL PROTECTED], Douglas Garstang [EMAIL PROTECTED] wrote: Peter, Too slow! We're going to potentially be doing several MySQL lookups for routing even the most basic of calls, and if every one of those queries has to make a call out to an AGI script, it would become a performance problem. AGI is only slow if you're calling it repeatedly and implementing it in a scripting language that needs a big interpreter. I have had great success writing AGIs in C and interfacing to MySQL from within them. They end up nice and small and fast. A single AGI invocation does all the database transactions necessary to decide on the disposition of the call, and then jumps to the appropriate extension and priority in the dialplan. And if you like writing your AGIs in interpreted languages, there is always FastAGI. ___ --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] Nested MySQL Commands
The most expensive part about perl is the init time. So you either want to find out how many calls per minute or most likely want FastAGI which eliminates this problem entirely. I wonder if FastAGI handles process pools, or if the FastAGIs need to handle sub-processes or threading. Douglas Garstang wrote: So I really wish there was some way to measure how well the worst case scenario would perform. This would be 120 simultaneous calls (don't know how many per second) on a Dual 3.8Ghz Dell PowerEdge 1850 with 2GB RAM. Asterisk would call an AGI script, written in perl, to route all calls. The script would have to perform multiple database queries in order to route a call. -Original Message- From: Mike Fedyk [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Nested MySQL Commands Chris Albertson wrote: Under Linux (and other OSes) It's not as bad as that. Even with 128 Perl processes running there is only one copy of the Perl interpeter in memory. Each of the 128 running processes would have it's own copy of only it's data segments. With Perl already in memory the biggest system overhead would be process creation. The best design is the one that minimizes the number of process that the kernel has to create. Notice that this is why the Apache Perl modual is so much faster than using Perl from a CGI script You will get the best usage of shared pages if all child interpreted processes fork off of one parent process. That way they can share as many data pages as possible also. If they don't fork off of each other, then a new copy of the interpreter will be put into memory. There will be some shared CoW pages between them, but not nearly as many when compared forking off of a common pool of processes. ___ --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] Re: Nested MySQL Commands
Yes, but the AGI process doesn't die, it stays running. If it stops for some reason, asterisk will start it again. http://www.voip-info.org/wiki-Asterisk+FastAGI http://www.sineapps.com/news.php?rssid=142 Douglas Garstang wrote: I don't get the whole concept of FastAGI. It's nothing special. Asterisk just opens a connection to a TCP port instead of executing a binary. -Original Message- From: Mike Fedyk [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Nested MySQL Commands Tony Mountifield wrote: In article [EMAIL PROTECTED], Douglas Garstang [EMAIL PROTECTED] wrote: Peter, Too slow! We're going to potentially be doing several MySQL lookups for routing even the most basic of calls, and if every one of those queries has to make a call out to an AGI script, it would become a performance problem. AGI is only slow if you're calling it repeatedly and implementing it in a scripting language that needs a big interpreter. I have had great success writing AGIs in C and interfacing to MySQL from within them. They end up nice and small and fast. A single AGI invocation does all the database transactions necessary to decide on the disposition of the call, and then jumps to the appropriate extension and priority in the dialplan. And if you like writing your AGIs in interpreted languages, there is always FastAGI. ___ --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] SoCal Users Group Meeting Schedule
Forwarded to OCLUG, LUGIE UUASC which have members that have expressed interest in asterisk. Mike Kerry Garrison wrote: The SoCal Asterisk Users Group will be meeting at the Heritage Park Public Library on the corner of Walnut and Yale in Irvine on the 3rd Thursday every month. The following dates are already secured: Thurs Jan 19 Thurs Feb 16 Thurs Mar 17 Irvine Heritage Park Library (949) 936-4040 14361 Yale Ave Irvine, CA 92604 Google Directions: http://tinyurl.com/9vq3e ___ --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] Still an open Seat in London for Next Weeks Signate intro to Asterisk Course
It might be easier to dig instead. [EMAIL PROTECTED] wrote: I would love to be there, but it's just too far to drive. regards, PaulH - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com; asterisk-biz@lists.digium.com Sent: Wednesday, January 11, 2006 7:24 AM Subject: [Asterisk-Users] Still an open Seat in London for Next Weeks Signate intro to Asterisk Course We still have a seat open in our Asterisk training course next week in London. You can find more information at our Web site, www.signate.com I'm going to be teaching the class. Paul ___ --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] Pri Gateway Hardware
Jean-Michel Hiver wrote: Alexander Lopez a écrit : TDMoE is stable and stale, There is no more development planed or needed as it only opens up a pipe between two ethernet points using Layer 2. OK... What would be in the advantage in running TDMoE rather than using IAX or SIP? TDMoE should allow for simpler firmware as it allows Asterisk to handle all of the details and just handles transferring the TDM data and failover in case of Asterisk server failure. It's about the same price as a TE411P and doesn't take a slot in your server(s). ___ --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] new AMPortal and Asterisk debs
Tzafrir Cohen wrote: On Sun, Jan 08, 2006 at 06:16:16PM -0800, Mike Fedyk wrote: Tzafrir Cohen wrote: Experimental: Asterisk 1.2: At the moment they are not that experimental anymore and should be ready for use, but are not well-tested yet. To use it, define both sources: deb http://rapid.dotsrc.org/ experimental/ How does this compare with Asterisk 1.2.1.dfsg-1 that is in etch/testing and 1.2.1.dfsg-3 that is in sid/unstable? Testing (Etch) is slightly behind. It is generally in line with the packages in Sid. Sort of. Actually ff you compare the changelog you'll find some striking similarities. In fact, it is based on the current version in the pkg-voip svn than to the current version. However it is built for Sarge (Stable). So if you have Sarge installed, you won't have to upgrade libc6/pgsql/pwdlib/whatever to use it. I try to commit most of the relevant changes and fixes to the main Debian package, so if you have Sid/Etch, you'll end up getting basically the same packages (only with a more changing base system...) . I didn't know you are a co-maintainer until now since I just checked. How much longer before this is on backports.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] Asterisk vs 3COM
Small, medium and large are relative. What do you want it to do, and why do you want to change your phone system? With the right talent, (consultant or in-house) Asterisk can be used in most situations. With that no more details, then a simple answer will have to suffice. Most likely yes. Dakota wrote: Would anyone recommend a medium size company choosing Asterisk over 3COM - Original Message - From: Kerry Garrison [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, January 07, 2006 10:23 PM Subject: RE: [Asterisk-Users] Asterisk Jobs If you try to compare Asterisk to other PBX's TODAY, Asterisk is running somewhere close to 0%. Its simply too new still as most companies didn't even begin taking a look until version 1.0 and even more with 1.2. Of course this will change over time. We are selling several systems a month right now. So if you are looking at getting a job today, it may be a little rough, but if you spend the next year honing your Asterisk skills more and more positions will open up. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Saturday, January 07, 2006 3:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Jobs I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places? - 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 ___ --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] Dialogic VFX/41JCT-LS found i a drawer
From: http://www.voip-info.org/wiki-Asterisk+Hardware Dialogic D/41JCT-LS Note: The D/41JCT-LS is a full duplex card and is the first of the D/41 family that will work with Asterisk. Older D/41 cards like D/41(E)PCI are half duplex cards designed for IVR type applications so they won't work for VoIP applications where you'd want to be able to to process both an incoming and outgoing stream simultaneouly. Note: Licenses for the D/41JCT-LS Asterisk drivers need to be purchased from Digium. Erick Perez wrote: I just found a Dialogic VFX/41JCT-LS (4 analog ports) in a drawer. I can use it in my house with asterisk at home project. Can I use that with asterisk? Where can I download proper drivers? -- --- 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
Re: [Asterisk-Users] new AMPortal and Asterisk debs
Tzafrir Cohen wrote: Experimental: Asterisk 1.2: At the moment they are not that experimental anymore and should be ready for use, but are not well-tested yet. To use it, define both sources: deb http://rapid.dotsrc.org/ experimental/ How does this compare with Asterisk 1.2.1.dfsg-1 that is in etch/testing and 1.2.1.dfsg-3 that is in sid/unstable? ___ --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 Gateway / Terminal for sale
Eric ManxPower Wieling wrote: JCC wrote: I don't get it. What is the advantage of using a GSM gateway? VOIP calls are pretty inexpensive as they are now. Is the use of a gateway intended as a backup incase a wired network connection goes down? I have being looking around the net for information on this. Anyone out there using it and if so you can please share with me how you use this technology? Any information will be appreciated. As you have seen from the other responses, there are advantages to this (and carriers support it) in other parts of the world. In the USA there isn't much need for such a device since calls to cell phones and calls to landlines cost the same amount. In other parts of the world a call to a cell phone is MUCH more expensive that calls to a landline. In many parts of Europe if you have a GSM gateway then the cell phones can become part of your Centrex system. They have their advantages in the US also. For instance, if you have your cell phones with a carrier that has free calls to the same network, then you can drastically reduce bills with a callback system. ___ --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] Processor Update?
Mike Hammett wrote: I've been Googling around for some time now (a few hours on dial-up). I find all kinds of questions similar to mine, but either there is no answer or the answer has nothing to do with the question. Hopefully this post isn't another one of those. Does Asterisk favor FPU performance or clock speed? (Meaning AMD or Intel) In this instance I think you're asking the wrong question. Asterisk will scale better on a processor that handles branching better, so the AMDs with the shorter pipelines will help here a lot. I see Asterisk can be compiled for x64 systems. Does it run any better\worse on x64 versus i386? I have heard good reports from x64 systems. Especially the AMD variety that isn't drain bamaged. Dual-core CPU performance isn't as good as dual CPU's, but is more often than not a better deal as it's close, but a lot cheaper. Dual-core vs. Dual CPU performance depends on the application. How does Asterisk respond? Looking to build out a system capable of 100 concurrent calls (IP pass-through, so in effect 200). Looking at the dimensioning, I can get some ideas, but few of them state the call quality. I'm hoping to be able to use a single CPU that is dual core as the price to go to a dual CPU that is single core puts me at least 60% to a redundant server. Asuming no AGI, a single p4 will handle 200 calls. More information about the setup -- No cards of any kind (Tyan board with integrated video, NIC, SATA nothing more needed) some IAX2 trunks (ztdummy) mainly SIP clients, but an increasing number of IAX2 mainly ulaw\alaw, if not solely (the transcoding chews CPU cycles needlessly) ___ --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] FastAGI available?
Is there anything like FastCGI for Asterisk so that AGIs can have persistent processes? ___ --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] Non-PRI T1
David Sampson wrote: Hello – I have a non-PRI T1 [...] How do I take incoming calls on these same channels? You should get a PRI T1. The minute you get close to capacity on this line you will run into timing issues with incoming and outgoing lines competing with each other. This problem will only happen when you need it most, which is the worst failure case. Unless your usage never spikes above 50% usage (counting incoming *and* outgoing) on this line you will regret not using a PRI. ___ --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] SIP permit/deny
Douglas Garstang wrote: I have the following in sip.conf. It was my understanding that this configuration (ie with deny/permit) would only allow connections from hosts 192.168.10.4 and 192.168.10.5. That doesn't seem to be the case. Asterisk is accepting INVITE's from other addresses. [a00090101] type=friend context=Company1 username=a00090101 ;secret=180 ;insecure=very host=dynamic [EMAIL PROTECTED] deny=0.0.0.0/0.0.0.0 permit=192.168.10.4/255.255.255.0 permit=192.168.10.5/255.255.255.0 Change your netmask to 255.255.255.255 or if possible use CIDR (/32). ___ --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: Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
Kevin P. Fleming wrote: If the two servers service distinctly separate groups of endpoints, they can share the same table since they won't care about the other server's entries. If the two servers service the same endpoints but in an active/passive arrangement, that would also work. Can the various *RT servers be configured to use different tables so there won't be any conflicts even if there is any client overlap between the servers? What I'm thinking of in this instance is active/active failover. Example: The HA system detects a peer has failed, fences it and then instructs asterisk to take over the registrations in table X that the failed peer was using. How close is this example to reality with *RT? ___ --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: Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
Kevin P. Fleming wrote: Mike Fedyk wrote: Can the various *RT servers be configured to use different tables so there won't be any conflicts even if there is any client overlap between the servers? Yes, but I'm not sure how you'd manage failover in that situation then. I was thinking of taking over the table when the other server fails and releasing once it is back up. What do you think? ___ --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] FC3 or FC1 (or something else?)
Brett, Gary wrote: From what ive read on this list and the wiki, centos 4.x has issues with the TE110P card ( a lot of people having issues after first reboot).Would 3.5 be better (I know [EMAIL PROTECTED] uses this) Am I right in saying that OS's with the 1.6 kernel still require a lot more tinkering than those with the 1.4 kernel ?? Does anybody know what Digiums stance on OS is , I remember speaking to them about 6 months ago and they were recommending a 1.4 kernel version of Debian. Are there any specific disadvantages to running 1.4 kernel ??, I presume you mean 2.4 and 2.6. Six months ago the Stable release of Debian couldn't run 2.6 kernels without installing a few updated packages from their backports.org repository. There has been a release since then that includes native 2.6 support. There are many areas where 2.6 improves upon 2.4 from processor and interrupt scalability to latency improvements. I would recommend any new server be installed with a 2.6 kernel unless there is some workload that requires a specific 2.4 kernel. I believe most of those were removed with the 2.6.5 to 2.6.8 anonVMA changes by Andrea. ___ --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 on Dell blade servers
Matt Riddell wrote: Alistair Cunningham wrote: We've been asked to quote for a large cluster running Asterisk and our ITSP in a box product. The system will be SIP throughout, with mixed codecs. We're considering using Dell blade servers, 1855 or similar, on the grounds that we normally use Dell machines and they work well, but we need higher rack density. Has anyone used these? Any feedback on whether they're good/bad/indifferent? What scalability do you get on simple SIP-SIP forwarding either with or without RTP passing through Asterisk? I would instead recommend the SuperMicro 1U servers - we have had a really great run with these. Do you use Opteron or Intel? ___ --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 termination services
If you are not big enough to have your own domain, then you don't need a disclaimer. This e-mail transmission may contain information that is non-proprietary, unprivileged and/or non-confidential and is intended exclusively to provide a clue to Jason D. Wolfe. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly allowed. If you are not the intended recipient or their designee, please distribute immediately so that people who try to wrap a contract around an insecure medium as a means of security will wisen up. Jason D. Wolfe wrote: I'm a newbie to Asterisk and telecom, and I I learned the hard way that analog POTS lines cause asterisk to start your dialplan as soon as the outbound starts ringing... that's why I was a little nervous about whether or not I may have the same problem using an IAX termination service. As it turns out, it works perfectly, as they do provide 'answer supervision' (like all digital lines). as well, I'm not going to erase my disclaimer below every time I send an email to a listserv. It does say 'person(s) to whom it is addressed', which keeps it from being completely senseless! :) and, I do NOT work for Bellsouth, they are my ISP... Jason Wolfe [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Jean-Michel Hiver Sent: Tuesday, January 03, 2006 1:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX termination services Jason D. Wolfe a crit : Hello, If I use an IAX termination service to connect outgoing VoIP calls to a PSTN will I have answer supervision so that my script won't initiate too early? I'm not sure to understand you. If you don't use Answer() before you use Dial(), asterisk won't answer until the dialed party does so. Cheers, Jean-Michel. ___ --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] M0n0Wall traffic shaping rules
Michael Graves wrote: On Wed, 04 Jan 2006 19:04:18 +0100, Matt Riddell wrote: I don't use m0n0wall, but wouldn't it be better just to shape based on a Type Of Service and then set the TOS flags in iax.conf and sip.conf accordingly? -- Cheers, Matt Riddell In a more general sense yes, TOS based QoS is better as it relates to outside your LAN. However, when using m0n0wall (great software!) it's easiest to assign priority based upon source machine (your * server) IP or port number. The examples given previously in this thread are derived from the built-in traffic shaping wizard. This establishes a series of weighted ques for data. All you really need to do is be certain that the IAX traffic is assigned to the highest priority que. Or all traffic to/from your server can be assigned to the hi priority que. It all sets up the same thing. Since QoS across the internet is pretty hard to achieve there's some question as to the actual usefullness of TOS bits. In future Telco/DSL providers may actually filter traffic looking for TOS tags to deter your from voip applications. Actually no, TOS is mostly useless because it has so few combinations, and you have to trust the sender to have a clue. Unless it originates from within your realm of control (most likely one of your LANs), then TOS is the last thing you should trust. It is only useful when you want to know the intent of the sender (which is seldom useful). That is why most examples use source and/or destination port specifications in addition to the IP addresses of the machines within your network(s) when assigning packet priority. This prevents people from taking advantage of your QoS rules. When you an ISP it is critical to think of these scenarios. It doesn't hurt to do it the same way in smaller setups where there is much less chance of someone trying to take advantage. Mike ___ --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
Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
Kevin P. Fleming wrote: Rich Adamson wrote: If you take the word dynamic out of that, then can he effectively have primary/secondary/backup systems that allows the user to re-register and/or redial his call on a different * server? I don't understand the question. I don't know if it was Rich's intention, but I'm interested in using RT for HA (High Availability) purposes. Think of this scenario: You have two * RT servers running heartbeat and one goes down. If the SIP registration information was kept in the DB tables, the backup server could take over the ethernet and IP addresses and continue without forcing the phones to re-register. ___ --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: Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
Kevin P. Fleming wrote: Mike Fedyk wrote: Think of this scenario: You have two * RT servers running heartbeat and one goes down. If the SIP registration information was kept in the DB tables, the backup server could take over the ethernet and IP addresses and continue without forcing the phones to re-register. Yes, that could work just as you described. With the current *RT release? ___ --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] FC3 or FC1 (or something else?)
Brett, Gary wrote: My question is which OS would be preferred in this configuration Fedora Core 1 or Fedora Core 3, and are there any install guides out there that are recent enough for asterisk 1.2 Use Debian or Centos (Free RHEL). ___ --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: What is the best Dell Machine for Asterisk?
Administrator TOOTAI wrote: Craig Guy a écrit : Are you using raid for performance or redundancy? Software raid is nice except when the drive that fails is the one with your boot partition on it. I guess you could always tftp boot the kernel or something. If you're using GRUB, fallback option allow you to boot on another boot partition if first failed. Yes, and if you don't get to GRUB, what do you do? I very much prefer a Linux software raid setup myself, but you are depending on the quirks of your BIOS if your primary boot drive dies. Mike ___ --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] What is the best Dell Machine for Asterisk?
Simone Cittadini wrote: Mike Fedyk ha scritto: Hiu Yen Onn wrote: How big of RAM for Asterisk server? My production environment will be about 400 users in the office. In one server? 4GB. And more if you can. I'd suggest you use several servers for 400 users unless the percentage of active phones is ~10%. Mike (with no agi and transcoding) 80 alaw concurrent calls , cdr_mysql, terminating on one TE410 Mem: 3105772k total, 733928k used, 2371844k free,8k buffers Cpu(s): 5.0% user, 5.5% system, 0.0% nice, 89.5% idle load average: 0.37, 0.39, 0.41 So that is ~80 calls per GB of ram which is 20% of 400 users so that should be 5 or 6GB to handle 100% usage. The load avg is the most important here. You want to keep it under 1.00 or you have processes waiting which increases jitter. Your system will be at 80% usage with 160 calls, assuming linear scaling. What are the specs for processor, memory and chipset that you pulled this stat from? ___ --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
