Re: [Asterisk-Users] Hard drive write cache
Hi, shadowym wrote: I am looking at ways to harden my asterisk install to prevent computer related issues from happening. I am concerned about about disk write cache. That seems to be a major source of hard drive corruption on power failure. Hard Drive corruption is simply unacceptable for the 99.999% uptime requirements of my Asterisk install that needs to be as reliable as a proprietary PBX. Things to consider: - Use compactflash to boot and run asterisk, add disk only for voicemail - Run the entire setup from a ram disk, make commit/rollback facilities to write to disk - Extra servers are cheap - you could use LinuxHA to failover the server. Florian ___ --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] Unable to connect to Asterisk? (simple[?] question)
SIP is a UDP protocol, and telnet is TCP. You can't test it like that. Have you tried connecting with a SIP client? Peter On 13/06/06, John Klimek [EMAIL PROTECTED] wrote: I'm trying to setup Asterisk on my Linksys WRT54G router and it appears to startup successfully (no errors) and it says it is listening on 0.0.0.0 port 5060, but I am unable to connect to it. I've tried telnet localhost 5060 but it just says connection refused. I've also tried connecting from another machine on my network (eg. telnet 192.168.0.1 5060) but it also says connection refused. Finally, I've tried changing the bound address in sip.conf to 127.0.0.1 and 192.168.0.1 but I am still unable to connect using all the methods mentioned above. What else can be the problem? Can I have some sort of iptables problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is Echo?
On Jun 12, 2006, at 10:04 PM, Crazy Boy wrote: Hi Friend, I heard about this word echo very much. Can you please tell me what is this Echo? Echo is when you say something and then hear it bounce back to you some brief time later... This can be caused by many things, but the most common in my opinion is cheap handsets and improper volumes. For example, you say hello Ashima into the phone, it travels across whatever technological monstrosity lies between you and the called party... When it get they hear your voice, but if the microphone on there handset hears it too, you might hear it again coming back to you. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and nortel meredian option 11c
Hi Koen Van Impe Thanks for the meridian config and asterisk. I will defenitly try them And let every one know. Just a few words and correct me if I am wrong There are two things 1 E1 : the 32 channels once both the equipment see each other and the ccs/hdb3 encoding/format is read the LED infront of interface goes green and this makes the lower layer work. 2 ISDN PRI: once step one is complete we can proceed to the signaling of ISDN PRI that is euro isdn or 5ess or any . I might be wrong But the problem that I face is the first step the e1 never comes up I have and the LED never goes green. I have checked the cable it work s fine with other pri which interms confirms the card also. But with the new config that u have given me I pray it works bcz it is very critical for my organization as we are tired of paying Nortel bags and bags of money and with this idea of using asterisk and interface it with the existing meridian system we see a hope of expanding with very little investment. Thanks and regards mohammad --- Best Regards Mohammad Zeeshan Latif Sr. WAN Engineer NETWORK DIRECTORATE 0092-51-90391020,0092-321-5181157 ___ --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] Bug in Voicemail ??
Hello, I have setup an Asterisk 1.2.7.1 system, with a working voicemail box: /etc/asterisk/extensions.conf exten = 83086921,1,Answer exten = 83086921,2,Dial(SIP/stefan,5,r) exten = 83086921,3,VoiceMail,u111 exten = 83086921,4,Hangup exten = 83086921,103,VoiceMail,b111 exten = 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 = 111,Mailbox 111,[EMAIL PROTECTED] The mailbox starts, I hear the intro and speak my message. In the CLI I can see that the message has been recorded and I get the recorded message via mail. But when I listen to the recorded messages or call the mailbox, I either hear nothing or just a short cracking sound. At least the length of the message is correct. If have tried to record the message with gsm, wav or wav49, the result is always the same. When I use the record() application to record a gsm file, everything is okay. I obviously made something wrong when configuring the voicemail system. Can someone give me a hint what's going wrong? Thanks for your help, stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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] How to retrieve voicemail
Hi, voicemail are working ok, I receive message as attach via email. My question is : how can the user call asterisk and listen to his voicemessages ? thanks Victor ___ --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] Bug in Voicemail ??
Hi, I'm still a newbie, but try to help you, my voicemail works ok, I can also record messages ok. My extension part is: exten = s,1,Background(welcome-cisl) exten = 1,1,Dial(Sip/vmoreno,10) exten = 1,2,Voicemail(victor) exten = 2,1,Dial(Sip/juliansip,10) exten = 2,2,Voicemail(aajulian) exten = 3,1,Playback(demo-echotest) exten = 3,2,Echo exten = 4,1,Congestion exten = 5,1,Dial(Sip/ludmila,10) exten = 5,2,Dial(Sip/vmoreno) exten = 6,1,Goto(testmenu,s,1) And voicemail.conf part is: [general] format=wav49 maxmessage=180 minmessage=2 maxsilence=2 silencethreshold=150 maxlogins=3 [EMAIL PROTECTED] skipms=3000 [victor] victor = 1234, Victor Moreno, [EMAIL PROTECTED] Hope it helps. One question to you, you say you call the malbox, how do you do that? which extension do i have to call to a ccess mailboxes? thank u Victor -- Victor Moreno CISL SPAIN, S.L. Parque Tecnológico de Andalucía Edif. Bic Euronova Avda. Juan López Peñalver, 21 29590 Campanillas (Málaga) Fax +34 95 10 10 561 Tlfn.: +34 95 10 10 581 Web: http://www.cisl.es Email: [EMAIL PROTECTED] Skype: victor.moreno Stefan-Michael. Guenther (in-put GbR) wrote: Hello, I have setup an Asterisk 1.2.7.1 system, with a working voicemail box: /etc/asterisk/extensions.conf exten = 83086921,1,Answer exten = 83086921,2,Dial(SIP/stefan,5,r) exten = 83086921,3,VoiceMail,u111 exten = 83086921,4,Hangup exten = 83086921,103,VoiceMail,b111 exten = 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 = 111,Mailbox 111,[EMAIL PROTECTED] The mailbox starts, I hear the intro and speak my message. In the CLI I can see that the message has been recorded and I get the recorded message via mail. But when I listen to the recorded messages or call the mailbox, I either hear nothing or just a short cracking sound. At least the length of the message is correct. If have tried to record the message with gsm, wav or wav49, the result is always the same. When I use the record() application to record a gsm file, everything is okay. I obviously made something wrong when configuring the voicemail system. Can someone give me a hint what's going wrong? Thanks for your help, stefan ___ --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 Echo?
Thank you Mr.Martin Joseph.Martin Joseph [EMAIL PROTECTED] wrote: On Jun 12, 2006, at 10:04 PM, Crazy Boy wrote: Hi Friend, I heard about this word "echo" very much. Can you please tell me what is this "Echo"?Echo is when you say something and then hear it bounce back to you some brief time later...This can be caused by many things, but the most common in my opinion is cheap handsets and improper volumes. For example, you say "hello Ashima" into the phone, it travels across whatever technological monstrosity lies between you and the called party... When it get they hear your voice, but if the microphone on there handset hears it too, you might hear it again coming back to you.Marty___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to retrieve voicemail
Victor Moreno wrote:
Hi,
voicemail are working ok, I receive message as attach via email.
My question is :
how can the user call asterisk and listen to his voicemessages ?
Set up a exten to voicemailmain passing the calling exten as the argument.
e.g.
exten = 121,1,VoiceMailMain(u${exten})
HTH
Jon
--
Jon Farmer
Telford, Shropshire, UK
___
--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] No reinvite - reason?
BJ Weschke schrieb: ... I have no modifiers in my dial command. ... One reason might be is if you are passing parameters in app_dial (eg. Hi, sorry, I did use the wrong expression. No, there is no parameter like tT in the Dial command. I think, I've made everything according to the docs. Anyway: No reinvite and no idea how to find the reason. Roger. ___ --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] How to retrieve voicemail
Hey. Maybe you can give me a hand with configuring my Linux box to send out emails? I've installed sendmail as per *several websites* and it's installed and running. I've gone into the voicemail.conf file and specified to allow attachments, etc. And, yes, I restarted Asterisk. Technically, I rebooted the entire Linux box. :) Anyway, any help would be appreciated. Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: Hi, voicemail are working ok, I receive message as attach via email. My question is : how can the user call asterisk and listen to his voicemessages ? thanks Victor ___ --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] Unicall acting really funny
Dear Moises, Thank you for your reply! I am not sure how to use the testcall tool to debug, but here we go with what I tried. The Main Thread message does never stop showing on the screen, I bet this is not the expected behavior tho. Maybe this can be of any help: asterisk-test:/usr/local/src/libunicall-0.0.3# cat testcall.conf destination-no 1150901010 protocol-class mfcr2 protocol-variant br,10,4 protocol-end cpe on-offered accept circuits 1-10 asterisk-test:/usr/local/src/libunicall-0.0.3# ./testcall Chan 1, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901010' Chan 2, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901011' Chan 3, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901012' Chan 4, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901013' Chan 5, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901014' Chan 6, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901015' Chan 7, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901016' Chan 8, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901017' Chan 9, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901018' Chan 10, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901019' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 Thread for channel 2 Thread for channel 3 Thread for channel 4 Thread for channel 5 Thread for channel 6 Thread for channel 7 Thread for channel 8 Thread for channel 9 Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Chan 3: -- Local end unblocked! :-) Chan 3: -- Local end unblocked! :-) Chan 4: -- Local end unblocked! :-) Chan 4: -- Local end unblocked! :-) Chan 5: -- Local end unblocked! :-) Chan 5: -- Local end unblocked! :-) Chan 6: -- Local end unblocked! :-) Chan 6: -- Local end unblocked! :-) Chan 7: -- Local end unblocked! :-) Chan 7: -- Local end unblocked! :-) Chan 8: -- Local end unblocked! :-) Chan 8: -- Local end unblocked! :-) Chan 9: -- Local end unblocked! :-) Chan 9: -- Local end unblocked! :-) Chan 10: -- Local end unblocked! :-) Chan 10: -- Local end unblocked! :-) Chan 1: -- Far end unblocked! :-) Chan 1: -- Far end unblocked! :-) Chan 1: Initiating call Chan 1: -- Dialing on channel 0 Chan 1: -- Dialing on channel 0 Chan 2: -- Far end unblocked! :-) Chan 2: -- Far end unblocked! :-) Chan 2: Initiating call Chan 2: -- Dialing on channel 0 Chan 2: -- Dialing on channel 0 Chan 3: -- Far end unblocked! :-) Chan 3: -- Far end unblocked! :-) Chan 3: Initiating call Chan 3: -- Dialing on channel 0 Chan 3: -- Dialing on channel 0 Chan 4: -- Far end unblocked! :-) Chan 4: -- Far end unblocked! :-) Chan 4: Initiating call Chan 4: -- Dialing on channel 0 Chan 4: -- Dialing on channel 0 Chan 5: -- Far end unblocked! :-) Chan 5: -- Far end unblocked! :-) Chan 5: Initiating call Chan 5: -- Dialing on channel 0 Chan 5: -- Dialing on channel 0 Chan 6: -- Far end unblocked! :-) Chan 6: -- Far end unblocked! :-) Chan 6: Initiating call Chan 6: -- Dialing on channel 0 Chan 6: -- Dialing on channel 0 Chan 7: -- Far end unblocked! :-) Chan 7: -- Far end unblocked! :-) Chan 7: Initiating call Chan 7: -- Dialing on channel 0 Chan 7: -- Dialing on channel 0 Chan 8: -- Far end unblocked! :-) Chan 8: -- Far end unblocked! :-) Chan 8: Initiating call Chan 8: -- Dialing on channel 0 Chan 8: -- Dialing on channel 0 Chan 9: -- Far end unblocked! :-) Chan 9: -- Far end unblocked! :-) Chan 9: Initiating call Chan 9: -- Dialing on channel 0 Chan 9: -- Dialing on channel 0 Chan 10: -- Far end unblocked! :-) Chan 10: -- Far end unblocked! :-) Chan 10: Initiating call Chan 10: -- Dialing on channel 0 Chan 10: -- Dialing on channel 0 Chan 3: -- Alerting on channel 0 Chan 3: -- Alerting on channel 0 Chan 8: -- Alerting on channel 0 Chan 8: -- Alerting on channel 0 Chan 1: -- Alerting on channel 0 Chan 1: -- Alerting on channel 0 Chan 4: -- Alerting on channel 0 Chan 4: -- Alerting on channel 0 Chan 2: -- Alerting on channel 0 Chan 2: -- Alerting on channel 0 Chan 6: -- Alerting on channel 0 Chan 6: -- Alerting on channel 0 Chan 7: -- Alerting on channel 0 Chan 7: -- Alerting on channel 0 Chan 9: -- Alerting on channel 0 Chan 9: -- Alerting on channel 0 Chan 5: -- Alerting on channel 0 Chan 5: -- Alerting on channel 0 Chan 10: -- Alerting on channel 0 Chan 10: -- Alerting on channel 0 Main thread Main thread Main thread Main thread Main thread Main thread Main thread Main thread Chan 6: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 6:
Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
Hi Josué,I just got the confirmation about integrating TE110P with TMS2 of Hipath 3750. Your help will be much appreciated.The configuration is as follow:PSTN - HIPATH 3750 (14 analog trunk lines) - TMS2 - TE110P - Asterisk All extensions of Hipath 3750 are analog (120 extensions)I know that it's maybe easier if we do other way, PSTN-ASterisk (2E-1) - TMS2 - Hipath 3750. But this is not an option, due to some political debat :((( I don't have the tech manual of Hipath yet, but here is what I want to do:1/ Calls from PSTN side arrived on Hipath 3750. Hipath somehow transfer that call into Asterisk box, using TMS2. Asterisk, functioning as an voicemail, feature server (voice log, conference, etc), after some menu prompts, will transfer back the call to Hipath 3750, using the same TMS2-TE110P connection, to one analog extension of Hipath 3750. 2/User of exteniosns of Hipath 3750, when dial out, will be transfered into Asterisk, using the same TMS2-TE110P. Asterisk will do the check of user balance account, LCR, and if approved , will transfer the call back to Hipath 3750, for getting into Analog trunk line. Since for the Hipath, TMS2 is a trunk module, so I suspect that some DISA operation must be enabled on Hipath, so we can enable the path from analog trunk port - TMS2 - Asterisk and back?Is above configuration working? And TMS2 use CAS, so do we have to use MFC/R2 (chan_unicall?)Very interested in your working configuration, can you explain a bit?Thank you and best regards,Nguyen On 5/26/06, Josué Conti [EMAIL PROTECTED] wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list thatmight help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a channelbank :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours: http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.htmlAnd this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to do whatever youwant. I don't know what DISA they are talking about? Do they mean S2Mor similar thing(but TMS2 is S2M)?Anyone?Sorry for not being able to help, but hope somebody else would do it.Benchev ___ --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] click to call features on asterisk
Firstly, thanks for the information, but I dont seem to get this SNAP work. I found out that the disadvantage of this is most computer dont come with mozilla, therefore for some non-IT literal is quite troublesome for them. Hmm..hopefully someone can provide me some info on click n call features. thanks in advance. On 6/10/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:You could check out Snap, there is a Firefox extension for it. You won't have to program webpages or anything as the phone numbers areautomatically detected and handled without needing anything extra fromthe web designer.http://www.snapanumber.com On 6/9/06, Sharon Lim [EMAIL PROTECTED] wrote: Hi colin, I am doing on php. But i would glad that you can share the codes as i will explore it. Thanks. On 6/9/06, Colin Anderson [EMAIL PROTECTED] wrote: I have, using Active Server Pages + Flash. See: http://new.landmarkmasterbuilder.com and click on Contact Call Us Online. I can post the .asp and .fla somewhere if someone is interested in it.-Original Message- From: Sharon Lim [mailto: [EMAIL PROTECTED]] Sent: Friday, June 09, 2006 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] click to call features on asterisk Hi there, anyone in the community has manage to configure click to call features? Care to share. I have tried on this manual , seem got some software error like http://www.voip-info.org/wiki/view/Asterisk+click+to+call Software error: Unable to determine call statusMessage: Originate with 'Exten' requires 'Context' and 'Priority' For help, please send mail to the webmaster ([EMAIL PROTECTED]), giving this error message and the time and date of the error. Thanks in advance. ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Bug in Voicemail ??
Hello Victor, Hi, I'm still a newbie, but try to help you, THX ;-)) And voicemail.conf part is: [general] format=wav49 maxmessage=180 minmessage=2 maxsilence=2 silencethreshold=150 maxlogins=3 [EMAIL PROTECTED] skipms=3000 [victor] victor = 1234, Victor Moreno, [EMAIL PROTECTED] Hope it helps. Thanks, I will compare it to my configuration. One question to you, you say you call the malbox, how do you do that? which extension do i have to call to a ccess mailboxes? You can define the extension as you like, here's my configuration: exten = 11101,1,Ringing exten = 11101,2,Wait(2) exten = 11101,3,VoiceMailMain,s111 exten = 22201,1,Ringing exten = 22201,2,Wait(2) exten = 22201,3,VoiceMailMain 11101 redirects your call directly to the mailbox 111, without asking for a password. 22201 wil ask you for the number of the mailbox and the password. Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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 SPA-941 NAT?
Very nice phones. There is no problem when conected to Asterisk (for about 6 months now) any body know this phone? support NAT? and standart codecs of asterisk ? -FD ___ --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] /var/log/asterisk/full ?
/etc/asterisk/logger.conf -FD Hi list! I have a Centos 4.3 box running Asterisk 1.2.9.1 with FreePBX 2.0.1 I noticed that this setup is keeping a full asterisk log which, after 1 month in production, has already grown to 1300 Mb in size. This is the log location : /var/log/asterisk/full Why is this on by default (I thought it is only used for debugging) and where can I disable it or at least have it rotated and gzipped like with other huge log files? 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 ___ --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] conference
Any one knows how to make a call conference using a voip gateway connected to asterisk. In mean what should I press (extension) to have another line and make the conference . regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] conference
Any one knows how to make a call conference using a voip gateway connected to asterisk. In mean what should I press (extension) to have another line and make the conference . regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
Hi I have next working sheme Hicom 350 - (Diun2 card)with DSS1- Asterisk with Quiad E1 This is work fine -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of NguyenSent: Tuesday, June 13, 2006 10:32 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750Hi Josué,I just got the confirmation about integrating TE110P with TMS2 of Hipath 3750. Your help will be much appreciated.The configuration is as follow:PSTN - HIPATH 3750 (14 analog trunk lines) - TMS2 - TE110P - Asterisk All extensions of Hipath 3750 are analog (120 extensions)I know that it's maybe easier if we do other way, PSTN-ASterisk (2E-1) - TMS2 - Hipath 3750. But this is not an option, due to some political debat :((( I don't have the tech manual of Hipath yet, but here is what I want to do:1/ Calls from PSTN side arrived on Hipath 3750. Hipath somehow transfer that call into Asterisk box, using TMS2. Asterisk, functioning as an voicemail, feature server (voice log, conference, etc), after some menu prompts, will transfer back the call to Hipath 3750, using the same TMS2-TE110P connection, to one analog extension of Hipath 3750. 2/User of exteniosns of Hipath 3750, when dial out, will be transfered into Asterisk, using the same TMS2-TE110P. Asterisk will do the check of user balance account, LCR, and if approved , will transfer the call back to Hipath 3750, for getting into Analog trunk line. Since for the Hipath, TMS2 is a trunk module, so I suspect that some DISA operation must be enabled on Hipath, so we can enable the path from analog trunk port - TMS2 - Asterisk and back?Is above configuration working? And TMS2 use CAS, so do we have to use MFC/R2 (chan_unicall?)Very interested in your working configuration, can you explain a bit?Thank you and best regards,Nguyen On 5/26/06, Josué Conti [EMAIL PROTECTED] wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list thatmight help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a "channelbank" :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours:http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.htmlAnd this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to do whatever youwant. I don't know what DISA they are talking about? Do they mean S2Mor similar thing(but TMS2 is S2M)?Anyone?Sorry for not being able to help, but hope somebody else would do it.Benchev ___ --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] timeout 't'
Hello, I have found that the timeout 't' takes to much time to be executed, around 10 seconds. Is there a place to configure this timeout ? thanks -- Victor Moreno CISL SPAIN, S.L. Parque Tecnológico de Andalucía Edif. Bic Euronova Avda. Juan López Peñalver, 21 29590 Campanillas (Málaga) Fax +34 95 10 10 561 Tlfn.: +34 95 10 10 581 Web: http://www.cisl.es Email: [EMAIL PROTECTED] Skype: victor.moreno ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Eyebeam chat function
Hi all, Eyebeam has a sip-chat function and it would be nice if I would be able to use it. But the problem is that I can't really find information about it. I can just try to send a message and on the Asterisk console a message like this appears: Jun 13 10:05:25 WARNING[6512]: chan_sip.c:7281 receive_message: Received message to sip:[EMAIL PROTECTED] from Bla Sheepsip:[EMAIL PROTECTED];tag=1d072048, dropped it... Content-Type:text/plain Message: ? Can anyone tell me more about this or give me a link with some information about it ? Regards, Attilla de GrootÎ ___ --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] Queues and macros and agents
When a caller in the queue is connected to an agent, the call is placed to the extension and context specified using Agentcallbacklogin. This allows for me to add extra things to the diaplan *before* calling the agent. Now, I want to be able to use a device, rather than agents. So I can use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device is called. Is there any way of doing that ? I was looking for something like the connected macro in the Dial command. I see that there is an optional agi command - I don't know agi, and will learn how to use it if required, but was hoping that there was something simple that I am missing. Julian ___ --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] voicemail suddenly exits on DTMF: a bug?
Hi, I'm using Asterisk 1.2.1 and I noticed that the voicemail suddenly exits if I press ANY key on the phone while the first or the second voice messages (es: vm-no.gsm or vm-youhave.gsm ) are played. I googled around but found nothing. How can I solve this problem? TIA Giorgio Incantalupo ___ --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] timeout 't'
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+timeout -FD Hello, I have found that the timeout 't' takes to much time to be executed, around 10 seconds. Is there a place to configure this timeout ? 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
[Asterisk-Users] VOCAL + Asterisk
I want to start a community based voip network projcet and am thinkimg of using VOCAL and asterisk gateways. my question is, has anyone bench marked asterisk vs VOCAL? is it a wise idea to use VOCAL + Asterisk or Asterisk all the way.am expecting 1000 - 5000 users.. your thoughts would be appreciated. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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
These days you don't have to worry much about your write cache unless you're running application where once single byte changed will affect whole file. Look at it this way, the only corruption will occur is whatever the files were open by asterisk at the time of the crash. And only up to the point where the file was last open. As far as I know asterisk does not keep cdr or log files open so you would loose only the data that was written at the time of the power failure. Any journaling file system (ext3, resierfs, xfs, etc) will easily handle any power failure event. Your files will not be corrupt but could miss some of the data. At the most you will loose 10-50 cdr entries written to you log files. If you post CDR to a remote SQL database then you asterisk install and linux is more or less static and will not be affected by the power failure. What you need to do is minimise the writes to hard disk's: 1 - Send syslog to remote server and do not do ANY syslogs Or keep the circular buffer in memory if you have plenty of it. 2 - Send CDR's to SQL server (or log to ramdisk and send to remote server every few minutes via SSH) 3 - Do not record any calls (or do that somewhere else) 4 - Stop any services that write/read data on regular intervals. If you have no writes you have nothing to worry about during power failure and journaling file system will take care of the rest. Keep your partition size really small so that fsck will not take much time. You have to be realistic, you cannot achieve 99.999% uptime. That's 5 minutes per year downtime. You will have more or less 100% until your first hardware failure. Even if you have all the hardware components pre-purchased it will still take you 2-12 hours to detect, diagnose and fix the fault if you lucky. So your 5 minuets If the business is demanding 99.999% then it should be prepared to invest into the hardware. I would recommend a cluster or even better a fault tolerant server. Those are expensive but you can pretty much rule out the hardware failure and swap all of the failed components while the system is running (cpu, memory, hdd, etc). Look at Stratus or NEC FT servers if you need hardware redundancy. They're expensive but will give you the hardware reliability you need. Or get a traditional PABX :) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of shadowym Sent: Tuesday, 13 June 2006 10:34 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hard drive write cache I am looking at ways to harden my asterisk install to prevent computer related issues from happening. I am concerned about about disk write cache. That seems to be a major source of hard drive corruption on power failure. Hard Drive corruption is simply unacceptable for the 99.999% uptime requirements of my Asterisk install that needs to be as reliable as a proprietary PBX. Of course I will be using redundant power supplies, raid 1 and use a UPS. None of those things mean much if the power cords accidentally get pulled from the back of the server. Unlikely as it may be I have to consider ALL possibilities. So is disabling the write cache a good way to reduce the risk of hard drive corruption for an Asterisk server? I am not too concerned about the reduced performance/lifetime of hardrives with write cache disabled since Asterisk is not a very write intensive environment. Even with lot's of voicemail going on. Any other recommendations/links for increasing the reliability of Asterisk servers? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: conference
Any one knows how to make a call conference using a voip gateway connected to asterisk. In mean what should I press (extension) to have another line and make the conference . regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
Hi, As long for HiPath 4000 callerID name doesn't work, only number -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pavel Jezek Sent: Thursday, May 25, 2006 9:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750 Hi, interoperability between asterisk and siemens interesting me too can you tell me, if caller id _name_ is fully working between asterisk and siemens, and what signaling do you use? currently I have Q.SIG signaling between siemens and ci$co voice gateway (with HDV-E1 module), but because ci$co can't decode caller id name from isdn/Q.SIG, I'm planning to replace ci$co gw with asterisk in near feature :-) PJ Josué Conti wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA2100 ringing without phone
Hello, We connected Sipura 2100 to Asterisk PBX. Plugged simple phone. Trying to call everything works ok. When we take out phone from Sipura, and trying to call, Asterisk shows, that Sipura is RINGING without phone connected to it. How could that be? -- SIP/240-2b03 is ringing How to tell SPA2100 to check for phone availability. Or is it just blind device which do not cares is phone connected to Line1 or not? Regards/Pagarbiai, Mindaugas Kezys ___ --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] VOCAL + Asterisk
Akpome Akpoguma a écrit : I want to start a community based voip network projcet and am thinkimg of using VOCAL and asterisk gateways. my question is, has anyone bench marked asterisk vs VOCAL? is it a wise idea to use VOCAL + Asterisk or Asterisk all the way.am expecting 1000 - 5000 users.. Apparently the way to do it is to use SER to handle all the SIP fluff (REGISTERs mostly) and then use Asterisk as a gateway for PSTN access. I used to do it but then realized that I didn't need it since I work mostly with wholesalers and don't have that much SIP signaling to deal with anyway. Then after that, you can use Asterisk simply as a B2BUA. As long as you don't do any transcoding, you will be fine. Currently I have 15-20k minutes daily going through an Asterisk box which just does two things: - Keep CDRs records in a database (using cdr_odbc) - Does dialplan functions (prefix manipulation, access control using contexts), load balancing (to balance traffic between multiple gateways, using Macros and Random()), and least cost routing. The machine is rather low-end (Sempron 2400+, 1 Gb RAM) but the load average is only about 0.2 and the CPU usage is around 10%. So for a low price tag of around €500 per unit, I can easily afford to have a second machine which can quickly take over if the main is down. All the transcoding and the echo cancellation is being handled by proprietary SIP VoIP gateways such as Audiocodes (excellent hardware, I recommend it). I use FreeBSD + Postgresql + unixODBC + Asterisk. I have set up a minimal Asterisk (use autoload = no in your modules.conf) and Asterisk's memory footprint is around 28M. The machine doesn't swap or hang... after a lot of research it seems I've found the combination which works for me! It took me two days + one sleepless night to set up though :) Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] conference
Any one knows how to make a call conference using a voip gateway connected to asterisk. In mean what should I press (extension) to have another line and make the conference . regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] conference
Have you sent this enough times yet? On 13/06/06, Khaled Chehab [EMAIL PROTECTED] wrote: Any one knows how to make a call conference using a voip gateway connected to asterisk. In mean what should I press (extension) to have another line and make the conference . regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
Hi Viktor,So where is the PSTN side on your schema? PSTN - Asterisk - Hicom 350? Or PSTN - Hicom 350- Asterisk?ThanksNguyenOn 6/13/06, Viktor Tatianin [EMAIL PROTECTED] wrote: Hi I have next working sheme Hicom 350 - (Diun2 card)with DSS1- Asterisk with Quiad E1 This is work fine -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of NguyenSent: Tuesday, June 13, 2006 10:32 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750Hi Josué,I just got the confirmation about integrating TE110P with TMS2 of Hipath 3750. Your help will be much appreciated.The configuration is as follow:PSTN - HIPATH 3750 (14 analog trunk lines) - TMS2 - TE110P - Asterisk All extensions of Hipath 3750 are analog (120 extensions)I know that it's maybe easier if we do other way, PSTN-ASterisk (2E-1) - TMS2 - Hipath 3750. But this is not an option, due to some political debat :((( I don't have the tech manual of Hipath yet, but here is what I want to do:1/ Calls from PSTN side arrived on Hipath 3750. Hipath somehow transfer that call into Asterisk box, using TMS2. Asterisk, functioning as an voicemail, feature server (voice log, conference, etc), after some menu prompts, will transfer back the call to Hipath 3750, using the same TMS2-TE110P connection, to one analog extension of Hipath 3750. 2/User of exteniosns of Hipath 3750, when dial out, will be transfered into Asterisk, using the same TMS2-TE110P. Asterisk will do the check of user balance account, LCR, and if approved , will transfer the call back to Hipath 3750, for getting into Analog trunk line. Since for the Hipath, TMS2 is a trunk module, so I suspect that some DISA operation must be enabled on Hipath, so we can enable the path from analog trunk port - TMS2 - Asterisk and back?Is above configuration working? And TMS2 use CAS, so do we have to use MFC/R2 (chan_unicall?)Very interested in your working configuration, can you explain a bit?Thank you and best regards,Nguyen On 5/26/06, Josué Conti [EMAIL PROTECTED] wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list thatmight help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a channelbank :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours:http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.html And this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to do whatever youwant. I don't know what DISA they are talking about? Do they mean S2Mor similar thing(but TMS2 is S2M)?Anyone?Sorry for not being able to help, but hope somebody else would do it.Benchev ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
Hi,Oh, I just want to get it to work. Caller Name is something luxurious for us .NguyenOn 6/13/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi,As long for HiPath 4000 callerID name doesn't work, only number -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Pavel Jezek Sent: Thursday, May 25, 2006 9:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750 Hi, interoperability between asterisk and siemens interesting me too can you tell me, if caller id _name_ is fully working between asterisk and siemens, and what signaling do you use? currently I have Q.SIG signaling between siemens and ci$co voice gateway (with HDV-E1 module), but because ci$co can't decode caller id name from isdn/Q.SIG, I'm planning to replace ci$co gw with asterisk in near feature:-) PJ Josué Conti wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] delay in MeetMe
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
RE: [Asterisk-Users] T1 passthrough/middleman
In zapata.conf I have, among other things:
; Incoming only
group=0 ; Zap/g0
signalling=pri_cpe
context=from-pstn
channel = 1-10
; Outgoing (only?)
group=1 ; Zap/g1
channel = 11-15,17-21
; To/From Alcatel
group=2 ; Zap/g2
signalling=pri_net
context=from-alcatel
channel = 32-46,48-62
Then in extensions.conf I have:
[from-pstn]
include = ext-did
include = from-pstn-timecheck ; this has to be included otherwise it
overrides ext-did
[ext-did]
; My DID has 3 numbers (scrambled to protect innocents): 987654ZXX
exten = _987654ZXX,1,Set(FROM_DID=_984899ZXX)
exten = _987654ZXX,2,Set(NumberCalled=${EXTEN:6})
exten = _987654ZXX,3,Goto(custom-ext-did,${EXTEN:6},1)
[custom-ext-did]
; use only 'include' here!!!
include = ext-local
; change trunk number below if trunks order changes!!!
include = outrt-003-alcatel
[from-pstn-timecheck]
...
(if incoming call doesn't match DID then do whatever you like...)
...
[outrt-003-alcatel]
; trunk '3' is Zap/g2 (To/From Alcatel)
exten = _ZXX,1,Macro(dialout-trunk,3,${EXTEN},)
exten = _ZXX,2,Macro(outisbusy); No available circuits
[from-alcatel]
; allow Alcatel phones to call Asterisk extensions
include = ext-local
; allow Alcatel phones to call PSTN numbers
include = from-alcatel-ext
exten = s,1,Macro(hangupcall)
exten = h,1,Macro(hangupcall)
[from-alcatel-ext]
exten = _X.,1,SetTransferCapability(SPEECH)
; trunk '2' is Zap/g1 (outgoing)
exten = _X.,n,Macro(dialout-trunk,2,${EXTEN},)
exten = _X.,n,Macro(outisbusy)
That's all (more or less)
Please send me a beer if it works for you!
Bye from Italy
M.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nathan Bell
Sent: Monday, June 12, 2006 6:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] T1 passthrough/middleman
That sounds exactly like what I want to do. I've don't have a
PRI line (although I'm going to press for getting one soon),
but for now I would just like a couple of pointers in getting
Asterisk's dial plan set up to just pass the calls from one
T1 to another.
Thanks a million in advance.
Mimmus wrote:
I used this approach to gradually migrate from a legacy Alcatel PBX:
PSTN E1 PRI --- Asterisk --- Crossed E1 cable --- Alcatel PBX At
first, Asterisk did nothing, only passing calls to/from Alcatel.
Then I started to use a bunch of SIP phones directly
connected to Asterisk.
Now I have the great part of extensions as SIP phones and
the old PBX
is working as a channel bank only for a few of analog devices.
Configuring the dialplan to do this dirty job is not
difficult but now
I'm not able to help you because it's saturday evening and
I'm at home!
Re-try next Monday.
DV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Nathan
Bell
Sent: Friday, June 09, 2006 10:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] T1 passthrough/middleman
Is it possible to act as a middle man on a T1 line?
My installation currently has an aging Inter-Tel Axxess box
with a T1
coming in (16 in, 8 out). Rather than adding and replacing
phones and
cards as they die, I would like to slowly migrate to a asterisk SIP
installation.
I want to take the incoming T1 line, use any available
outgoing lines
for outgoing SIP, intercept any incoming lines and either send them
off to a SIP line or pass them through to other T1 line
(going to the
Axxess box), and finally take in outgoing calls from the
Inter-Tel box
and either send them to SIP or send them to the outside T1 line.
How will a dual T1 card be set up in this situation? Would it be
easier to use an FXO channel bank (or card) and connect
analog lines
to the FXS analog lines on the Inter-Tel box?
___
--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] conference
each ip phone need to register to be able to call the conference. Firstly, you need to create user with username and password under sip.conf, Then you need to create conference room which is meetme in meetme.conf then you need to create extension to point to the conference room in extensions.confafter that, just click on the extension that refer in conference room. for more info, read this http://www.voip-info.org/wiki-Asterisk+cmd+MeetMehope this help. On 6/13/06, Peter Bowyer [EMAIL PROTECTED] wrote:Have you sent this enough times yet?On 13/06/06, Khaled Chehab [EMAIL PROTECTED] wrote: Any one knows how to make a call conference using a voip gateway connected to asterisk. In mean what should I press (extension)to have another line and make the conference . regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Peter BowyerEmail: [EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI Broke on 1.2.9.1?
Hi Everyone,
This morning I upgraded Asterisk from 1.2.7.1 to 1.2.9.1 along with libpri to version 1.2.3 and Zaptel to 1.2.6. Then stopped and restarted all the Asterisk and Zaptel components. Service resumed as usual but after about 30minutes the console was filled with errors and the PRI ZAP channel (Digium TDM 2 Span) stopped answering calls. The errors where as follows:
pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner
Forcing restart of channel 0/9 on span 2 since channel reported in use
Channel 0/9, span 2 received AOC-E charging 0 units
channel.c:787 channel_find_locked: Avoided initial deadlock for '0x74c570', 10 retries!
At first I thought it might be a problem with my telco provider (BT) but I decided to downgrade to 1.2.7.1 and everything works fine like it did before the upgrade. Im just wondering if something to do with PRI / Zaptel has been broken in the update?. Any Ideas? I really need this update as it appears to fix a bug that I have with IAX2 which stops listening to frames and timeouts all calls even when the latency is averaging 5ms.
Thanks
Chris Teesdale
I.T. Development / I.P Telephony Development
Philips
Tel : 01325 384394 ex 221
Email : [EMAIL PROTECTED]
IMPORTANT:
This email and any attachments may be confidential and/or privileged. Everything is intended for use of the addressee only. If you are not the
named addressee you must not disseminate, distribute or copy this email. If you receive this email in error please notify the sender by replying
to this email or by telephoning (+44)(0)870 609 1554 then delete this message from your system. Philips Collection Services Ltd. ("Philips")
routinely monitors the content of email sent and received on its network, to ensure compliance with its policies and procedures. Although
Philips have taken reasonable precautions to ensure no viruses are present in this email or any files attached to it, it cannot accept
any responsibility for any loss or damage arising from the use of this email or its attachments and advises you to carry out appropriate
virus checks. Philips are not responsible for any changes made to the message after it has been sent nor any files attached to it after
it was sent. Emails that contain encrypted material, program files, are obscene, inflammatory, criminal, offensive, in breach of copyright,
contain a virus or threat to computer systems, appear to be a threat to the company or in breach of company policy may be intercepted and/or deleted.
Philips does not accept any liability for any statements made which are clearly the sender's own and not made on behalf of Philips.
___
--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] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06
Hi, I made the mistake of upgrading both my Linux box (to Ubuntu 6.06) and Asterisk (to 1.2.9.1) at the same time. Now, when trying to compile mpg123 - using the tried and true make mpg123 -, the build fails with an error make[3]: Entering directory `/usr/src/asterisk-1.2.9.1/mpg123-0.59r' make[3]: *** No rule to make target `\ ', needed by `mpg123'. Stop. Maybe there's someone out there more versed in Linux who has an idea what might have gone wrong. Thanks! Marc ___ --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] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06
Try make on its own and read what it says. You probably want make linux Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc Rohlfing Sent: 13 June 2006 12:09 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06 Hi, I made the mistake of upgrading both my Linux box (to Ubuntu 6.06) and Asterisk (to 1.2.9.1) at the same time. Now, when trying to compile mpg123 - using the tried and true make mpg123 -, the build fails with an error make[3]: Entering directory `/usr/src/asterisk-1.2.9.1/mpg123-0.59r' make[3]: *** No rule to make target `\ ', needed by `mpg123'. Stop. Maybe there's someone out there more versed in Linux who has an idea what might have gone wrong. Thanks! Marc ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues and macros and agents
On 6/13/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: When a caller in the queue is connected to an agent, the call is placed to the extension and context specified using Agentcallbacklogin. This allows for me to add extra things to the diaplan *before* calling the agent. Now, I want to be able to use a device, rather than agents. So I can use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device is called. Is there any way of doing that ? I was looking for something like the connected macro in the Dial command. I see that there is an optional agi command - I don't know agi, and will learn how to use it if required, but was hoping that there was something simple that I am missing. If you're not using agents, the AGI will be your only shot, but that's also executed against the calling channel and not the device you've called who has the agent on the other end. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI Broke on 1.2.9.1?
I have been running the same software versions together with a digium single port PRI card in the UK and have not experienced any problems since the upgrade. On Tue, 2006-06-13 at 12:00, Chris Teesdale wrote: Hi Everyone, This morning I upgraded Asterisk from 1.2.7.1 to 1.2.9.1 along with libpri to version 1.2.3 and Zaptel to 1.2.6. Then stopped and restarted all the Asterisk and Zaptel components. Service resumed as usual but after about 30minutes the console was filled with errors and the PRI ZAP channel (Digium TDM 2 Span) stopped answering calls. The errors where as follows: pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner Forcing restart of channel 0/9 on span 2 since channel reported in use Channel 0/9, span 2 received AOC-E charging 0 units channel.c:787 channel_find_locked: Avoided initial deadlock for '0x74c570', 10 retries! At first I thought it might be a problem with my telco provider (BT) but I decided to downgrade to 1.2.7.1 and everything works fine like it did before the upgrade. Im just wondering if something to do with PRI / Zaptel has been broken in the update?. Any Ideas? I really need this update as it appears to fix a bug that I have with IAX2 which stops listening to frames and timeouts all calls even when the latency is averaging 5ms. Thanks Chris Teesdale I.T. Development / I.P Telephony Development Philips® Tel : 01325 384394 ex 221 Email : [EMAIL PROTECTED] __ IMPORTANT: This email and any attachments may be confidential and/or privileged. Everything is intended for use of the addressee only. If you are not the named addressee you must not disseminate, distribute or copy this email. If you receive this email in error please notify the sender by replying to this email or by telephoning (+44)(0)870 609 1554 then delete this message from your system. Philips Collection Services Ltd. (Philips) routinely monitors the content of email sent and received on its network, to ensure compliance with its policies and procedures. Although Philips have taken reasonable precautions to ensure no viruses are present in this email or any files attached to it, it cannot accept any responsibility for any loss or damage arising from the use of this email or its attachments and advises you to carry out appropriate virus checks. Philips are not responsible for any changes made to the message after it has been sent nor any files attached to it after it was sent. Emails that contain encrypted material, program files, are obscene, inflammatory, criminal, offensive, in breach of copyright, contain a virus or threat to computer systems, appear to be a threat to the company or in breach of company policy may be intercepted and/or deleted. Philips does not accept any liability for any statements made which are clearly the sender's own and not made on behalf of Philips. __ __ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Can this config sustain 30 users?
EP == Erick Perez [EMAIL PROTECTED] writes: EP BJ, when you say it is more than adequate, what do you do to EP calculate? there *must* be a way to at least tell if the EP motherboardboard/cpu will achieve results. I just don't want to EP install it and then after a 5th user going to call someone the EP asterisk begin to crash due to lack of resuources. Our E1 boxes are only slightly beefier and they report 98%-100% idle in vmstat. Right now there's 16 calls on one of them and it's 99% idle. We don't transcode though, everything is Alaw/ulaw. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Realtime and Ex-Girlfriend
Hallo all,
Last night I have successfully setup Asterisk Realtime with mysql. but I
have one problem regarding the Ex-Girlfrind-Functionality.
The example: I have a fax running on a specific extension (300) and I
want that one to call out via ISDN, but it simply gets IGNORED.
I have tried _X./300 = Dial(zap/g1/${EXTEN})
but what happens now is that (because of includes defined) all other
calls are using this extension, even if somebody completely different
(for example 55) wants to call outside.
The dial itself works, but i simply just want it running as usual in
extensions.conf. I now (temporarily) have made a GotoIf function before
the standard dial, but that definitly isnt a very nice option.
Help in this manner would be greatly appreciated.
Michael Kromer
CC Computer Consultants GmbH
___
--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] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06
Why still use mpg123? Start using format_mp3 from asterisk-addons and your * will play mp3 by itself... K On 6/13/06, Marc Rohlfing [EMAIL PROTECTED] wrote: Hi,I made the mistake of upgrading both my Linux box (to Ubuntu 6.06) andAsterisk (to 1.2.9.1) at the same time. Now, when trying to compilempg123 - using the tried and true make mpg123 -, the build fails withan errormake[3]: Entering directory `/usr/src/asterisk- 1.2.9.1/mpg123-0.59r'make[3]: *** No rule to make target `\', needed by `mpg123'.Stop.Maybe there's someone out there more versed in Linux who has an ideawhat might have gone wrong. Thanks! Marc___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
I use PSTN - Hicom 350- Asterisk Asterisk I use for voice mail, ivr and gateway for voice overip I try connect Asterisk to PSTN with EDSS1 signaling it work fine at PSTN side statioon type 5ESS What problem you have ? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of NguyenSent: Tuesday, June 13, 2006 1:37 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750Hi Viktor,So where is the PSTN side on your schema? PSTN - Asterisk - Hicom 350? Or PSTN - Hicom 350- Asterisk?ThanksNguyen On 6/13/06, Viktor Tatianin [EMAIL PROTECTED] wrote: Hi I have next working sheme Hicom 350 - (Diun2 card)with DSS1- Asterisk with Quiad E1 This is work fine -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of NguyenSent: Tuesday, June 13, 2006 10:32 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750Hi Josué,I just got the confirmation about integrating TE110P with TMS2 of Hipath 3750. Your help will be much appreciated.The configuration is as follow:PSTN - HIPATH 3750 (14 analog trunk lines) - TMS2 - TE110P - Asterisk All extensions of Hipath 3750 are analog (120 extensions)I know that it's maybe easier if we do other way, PSTN-ASterisk (2E-1) - TMS2 - Hipath 3750. But this is not an option, due to some political debat :((( I don't have the tech manual of Hipath yet, but here is what I want to do:1/ Calls from PSTN side arrived on Hipath 3750. Hipath somehow transfer that call into Asterisk box, using TMS2. Asterisk, functioning as an voicemail, feature server (voice log, conference, etc), after some menu prompts, will transfer back the call to Hipath 3750, using the same TMS2-TE110P connection, to one analog extension of Hipath 3750. 2/User of exteniosns of Hipath 3750, when dial out, will be transfered into Asterisk, using the same TMS2-TE110P. Asterisk will do the check of user balance account, LCR, and if approved , will transfer the call back to Hipath 3750, for getting into Analog trunk line. Since for the Hipath, TMS2 is a trunk module, so I suspect that some DISA operation must be enabled on Hipath, so we can enable the path from analog trunk port - TMS2 - Asterisk and back?Is above configuration working? And TMS2 use CAS, so do we have to use MFC/R2 (chan_unicall?)Very interested in your working configuration, can you explain a bit?Thank you and best regards,Nguyen On 5/26/06, Josué Conti [EMAIL PROTECTED] wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list thatmight help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a "channelbank" :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours:http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.html And this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to
Re: [Asterisk-Users] Unable to connect to Asterisk? (simple[?] question)
Ahhh, that would explain it. I setup my firewall (eg. Shorewall) to allow incoming TCP connections to port 5060. I've changed it to UDP port 5060 and it works great! (well, Asterisk says Forbidden, but that's just a simple config problem I'm sure) Which other ports do I need to forward/open to get Asterisk working properly? I'm guessing I only need port 5060 open to my local network... Do I need any ports open for connections from the internet? (eg. incoming connections) On 6/13/06, Peter Bowyer [EMAIL PROTECTED] wrote: SIP is a UDP protocol, and telnet is TCP. You can't test it like that. Have you tried connecting with a SIP client? Peter On 13/06/06, John Klimek [EMAIL PROTECTED] wrote: I'm trying to setup Asterisk on my Linksys WRT54G router and it appears to startup successfully (no errors) and it says it is listening on 0.0.0.0 port 5060, but I am unable to connect to it. I've tried telnet localhost 5060 but it just says connection refused. I've also tried connecting from another machine on my network (eg. telnet 192.168.0.1 5060) but it also says connection refused. Finally, I've tried changing the bound address in sip.conf to 127.0.0.1 and 192.168.0.1 but I am still unable to connect using all the methods mentioned above. What else can be the problem? Can I have some sort of iptables problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [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
Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
Hi Nguyen. The Asterisk with configured 50 SIP Phones is integrated with HiPath 3750, through a board TMS2. For access the PSTN I created a context in asterisk so that asterisk has access HiPath 3750 and uses the LCR's that I configured in the HiPath. The CDR's do asterisk is registered no billing do HiPath that is the Siemens Call Report. The system of voicemail for all the set (HiPath 3750 and Asterisk) I use the Asterisk with access to the Internet, where all the 120 users receive its messages perfectly and also they receive copy from the message for email. The interconnection that we effect is ETSI or EuroISDN. I wait to have helped.If to need plus some thing is to inform. Best Regards Josué 2006/6/13, Viktor Tatianin [EMAIL PROTECTED]: I use PSTN - Hicom 350- Asterisk Asterisk I use for voice mail, ivr and gateway for voice overip I try connect Asterisk to PSTN with EDSS1 signaling it work fine at PSTN side statioon type 5ESS What problem you have ? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Nguyen Sent: Tuesday, June 13, 2006 1:37 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750 Hi Viktor,So where is the PSTN side on your schema? PSTN - Asterisk - Hicom 350? Or PSTN - Hicom 350- Asterisk?ThanksNguyen On 6/13/06, Viktor Tatianin [EMAIL PROTECTED] wrote: Hi I have next working sheme Hicom 350 - (Diun2 card)with DSS1- Asterisk with Quiad E1 This is work fine -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ]On Behalf Of NguyenSent: Tuesday, June 13, 2006 10:32 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750 Hi Josué,I just got the confirmation about integrating TE110P with TMS2 of Hipath 3750. Your help will be much appreciated.The configuration is as follow:PSTN - HIPATH 3750 (14 analog trunk lines) - TMS2 - TE110P - Asterisk All extensions of Hipath 3750 are analog (120 extensions)I know that it's maybe easier if we do other way, PSTN-ASterisk (2E-1) - TMS2 - Hipath 3750. But this is not an option, due to some political debat :((( I don't have the tech manual of Hipath yet, but here is what I want to do:1/ Calls from PSTN side arrived on Hipath 3750. Hipath somehow transfer that call into Asterisk box, using TMS2. Asterisk, functioning as an voicemail, feature server (voice log, conference, etc), after some menu prompts, will transfer back the call to Hipath 3750, using the same TMS2-TE110P connection, to one analog extension of Hipath 3750. 2/User of exteniosns of Hipath 3750, when dial out, will be transfered into Asterisk, using the same TMS2-TE110P. Asterisk will do the check of user balance account, LCR, and if approved , will transfer the call back to Hipath 3750, for getting into Analog trunk line. Since for the Hipath, TMS2 is a trunk module, so I suspect that some DISA operation must be enabled on Hipath, so we can enable the path from analog trunk port - TMS2 - Asterisk and back?Is above configuration working? And TMS2 use CAS, so do we have to use MFC/R2 (chan_unicall?)Very interested in your working configuration, can you explain a bit?Thank you and best regards,Nguyen On 5/26/06, Josué Conti [EMAIL PROTECTED] wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list that might help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a channelbank :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours: http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.html And this proves that the idea must work. - When the call arrived from PSTN through CO line,
Re: [Asterisk-Users] Unable to connect to Asterisk? (simple[?] question)
Try this: http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules Peter On 13/06/06, John Klimek [EMAIL PROTECTED] wrote: Ahhh, that would explain it. I setup my firewall (eg. Shorewall) to allow incoming TCP connections to port 5060. I've changed it to UDP port 5060 and it works great! (well, Asterisk says Forbidden, but that's just a simple config problem I'm sure) Which other ports do I need to forward/open to get Asterisk working properly? I'm guessing I only need port 5060 open to my local network... Do I need any ports open for connections from the internet? (eg. incoming connections) On 6/13/06, Peter Bowyer [EMAIL PROTECTED] wrote: SIP is a UDP protocol, and telnet is TCP. You can't test it like that. Have you tried connecting with a SIP client? Peter On 13/06/06, John Klimek [EMAIL PROTECTED] wrote: I'm trying to setup Asterisk on my Linksys WRT54G router and it appears to startup successfully (no errors) and it says it is listening on 0.0.0.0 port 5060, but I am unable to connect to it. I've tried telnet localhost 5060 but it just says connection refused. I've also tried connecting from another machine on my network (eg. telnet 192.168.0.1 5060) but it also says connection refused. Finally, I've tried changing the bound address in sip.conf to 127.0.0.1 and 192.168.0.1 but I am still unable to connect using all the methods mentioned above. What else can be the problem? Can I have some sort of iptables problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 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
[Asterisk-Users] Re: Linksys SPA-941 NAT?
On Tue, Jun 13, 2006 at 09:53:36AM +0200, Filip Dr?gowski wrote: Very nice phones. There is no problem when conected to Asterisk (for about 6 months now) any body know this phone? support NAT? and standart codecs of asterisk ? thank you all!! -FD ___ --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 ---end quoted text--- -- .- ___ --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 Eyebeam chat function
Attilla De Groot wrote: Hi all, Eyebeam has a sip-chat function and it would be nice if I would be able to use it. But the problem is that I can't really find information about it. I can just try to send a message and on the Asterisk console a message like this appears: Jun 13 10:05:25 WARNING[6512]: chan_sip.c:7281 receive_message: Received message to sip:[EMAIL PROTECTED] from Bla Sheepsip:[EMAIL PROTECTED];tag=1d072048, dropped it... Content-Type:text/plain Message: ? Can anyone tell me more about this or give me a link with some information about it ? Far as I know, Asterisk doesn't support the SIMPLE SIP text protocol. You should use a full-function SIP proxy like SER. ___ --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] Fun with Echo
Andrew Kohlsmith wrote: On Monday 12 June 2006 17:55, shadowym wrote: Believe me, you can drive yourself insane trying to come up with some magical formula that JUST works because it usually won't happen that way. Software echo cancellers are simply not good enough for many situations. Actually that's untrue. I think (hope) that Steve Underwood will jump in here and correct me, but it's my understanding that the only real reason why the software echo cancellers available in Zaptel don't work as well as the hardware echo cancellers from Tellabs and the Octasic chips in the Sangoma and Digium hardware echo cancellers is because of implementation. There is a spec for echo cancellation on PSTN called g.168. I believe it's a suite of tests which put the echo canceller through its paces and if you pass them you are certified to conform to g.168. None of the echo cancellers in zaptel conform to this, whereas the Octasic, Tellabs and other hardware echo cancellers all do. If someone were to put the effort and energy into making the software echo cancellers compliant, you should find similar results to the hardware echo cans. The echo cancellers in Zaptel are far better than anything I could throw together myself, and there's a lot of heavy math and dark juju hiding inside that optimized code, but they're all still very much proof of concept and test code compared to a true g.168-compliant echo can. Basically they're there for free and might get you what you need, but they're certainly not a reflection of all that is possible with a general CPU echo canceller. Since you invited me, see http://www.soft-switch.org/dumb-vs-smart/ar01.html Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream GXV-3000
Mike Fedyk wrote: Or any polycom phone that has speakerphone like the IP501 and IP430. Time Bandit wrote: Can you, or anyone else comment on the speakerphone ability of the GVX-3000 ? We run the GXP-2000's and for the most part are happy with them, but for upper management we're looking at phones with better speakerphone. These would be ideal if the speakerphone isn't as terrible as the GXP-2000. I never tried the GVX-3000, but I can recommend with confidence a Cisco 7940 or 7960 for the quality of the speakerphone. And how good do you find the video on the Cisco 7940, 7960 and Polycom IP501, IP430? Personally, I've never seen them do video, but if you are suggesting them as a substitute for a GVX-3000 I guess I must be wrong. :-) Steve ___ --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] Unicall acting really funny
Joao. The more important thing about testcall is that it allows you to isolate the problem (you dont need asterisk) . I would recommend you to read this document I wrote: http://phpmexic.u33.0web-hosting.com/wordpress/misc/mfcr2-asterisk-unicall.pdf Here is described a bit more detailed. Is in spanishg, but since im able to read portuguese I think may be you can understand it. The more important thing is to increment testcall logging verbosity. For that you need to edit testcall.c, there you will find a function named run_uc(), search the logging_level variable and let it be UC_LOG_SEVERITY_MASK, then make testcall and run the test again but use only in 1 circuit. That will show in wich part of the initial signaling is staying stuck. Other more clear test is modify mfcr2.c, in the first line it has commented a #define AUDIO_LOG, uncomment it and libmfcr2 will save all the audio signalingin in a couple of files in the same folder you execute testcall. ONce done that, you can send me the audio files and the logging messages and may be I will be able to know what the problem is. Regards On 6/13/06, Joao Mesquita [EMAIL PROTECTED] wrote: Dear Moises, Thank you for your reply! I am not sure how to use the testcall tool to debug, but here we go with what I tried. The Main Thread message does never stop showing on the screen, I bet this is not the expected behavior tho. Maybe this can be of any help: asterisk-test:/usr/local/src/libunicall-0.0.3# cat testcall.conf destination-no 1150901010 protocol-class mfcr2 protocol-variant br,10,4 protocol-end cpe on-offered accept circuits 1-10 asterisk-test:/usr/local/src/libunicall-0.0.3# ./testcall Chan 1, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901010' Chan 2, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901011' Chan 3, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901012' Chan 4, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901013' Chan 5, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901014' Chan 6, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901015' Chan 7, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901016' Chan 8, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901017' Chan 9, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901018' Chan 10, class 'mfcr2', variant 'br,10,4', end 1, caller 1461727523, from '' to '1150901019' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 Thread for channel 2 Thread for channel 3 Thread for channel 4 Thread for channel 5 Thread for channel 6 Thread for channel 7 Thread for channel 8 Thread for channel 9 Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Chan 3: -- Local end unblocked! :-) Chan 3: -- Local end unblocked! :-) Chan 4: -- Local end unblocked! :-) Chan 4: -- Local end unblocked! :-) Chan 5: -- Local end unblocked! :-) Chan 5: -- Local end unblocked! :-) Chan 6: -- Local end unblocked! :-) Chan 6: -- Local end unblocked! :-) Chan 7: -- Local end unblocked! :-) Chan 7: -- Local end unblocked! :-) Chan 8: -- Local end unblocked! :-) Chan 8: -- Local end unblocked! :-) Chan 9: -- Local end unblocked! :-) Chan 9: -- Local end unblocked! :-) Chan 10: -- Local end unblocked! :-) Chan 10: -- Local end unblocked! :-) Chan 1: -- Far end unblocked! :-) Chan 1: -- Far end unblocked! :-) Chan 1: Initiating call Chan 1: -- Dialing on channel 0 Chan 1: -- Dialing on channel 0 Chan 2: -- Far end unblocked! :-) Chan 2: -- Far end unblocked! :-) Chan 2: Initiating call Chan 2: -- Dialing on channel 0 Chan 2: -- Dialing on channel 0 Chan 3: -- Far end unblocked! :-) Chan 3: -- Far end unblocked! :-) Chan 3: Initiating call Chan 3: -- Dialing on channel 0 Chan 3: -- Dialing on channel 0 Chan 4: -- Far end unblocked! :-) Chan 4: -- Far end unblocked! :-) Chan 4: Initiating call Chan 4: -- Dialing on channel 0 Chan 4: -- Dialing on channel 0 Chan 5: -- Far end unblocked! :-) Chan 5: -- Far end unblocked! :-) Chan 5: Initiating call Chan 5: -- Dialing on channel 0 Chan 5: -- Dialing on channel 0 Chan 6: -- Far end unblocked! :-) Chan 6: -- Far end unblocked! :-) Chan 6: Initiating call Chan 6: -- Dialing on channel 0 Chan 6: -- Dialing on channel 0 Chan 7: -- Far end unblocked! :-) Chan 7: -- Far end unblocked! :-) Chan 7: Initiating call Chan 7: -- Dialing on channel 0 Chan 7: -- Dialing on channel 0 Chan 8: -- Far end unblocked! :-) Chan 8: -- Far end unblocked! :-) Chan 8: Initiating call Chan 8: -- Dialing on channel 0 Chan 8: -- Dialing on channel 0 Chan 9: -- Far end unblocked! :-) Chan 9: -- Far end unblocked! :-) Chan 9:
[Asterisk-Users] Asterisk and TBCT
Hi Guys, I'm starting to work on Asterisk, trying to see if it will fit our needs, but so far it seems it doesn't support TBCT(Two B Channel Transfer). I've found a couple of links that was talking about TBCT, and someone had posted a bounty for that feature, but no news since 2003. I've also seen that it seems there is some support in libpri, or work was done around 2005 for TBCT and there's a mention about supporting TBCT but only with 5ESS. And at the moment I don't have the equipement to try it out, so before we do try it out I was trying to investigate a bit. In the case that there's no support for TBCT, is there other possible way we can do it ? 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] Xorcom Rapid
Hello,
I've receive no response, no idea??
Bets regards,
Olivier S;
Tzafrir Cohen a écrit :
Jut as usual with Zaptel: Zap/NNN (e.g: Zap/1 , Zap/2) for individual
channels. And gNNN and similar work just the same.
OK, in extensions.conf, i put the contexts PSTN and INTERNAL as:
[PSTN] ; for in coming calls - defin in zapata.conf
exten = s,1,Dial(IAX2/300,20)
exten = s,2,Voicemail, u300)
[INTERNAL] ; for internal AND outgoing call - actually just outgoing calls
exten = _0.,1,Dial(ZAP/g1/${EXTEN:1})
For hardware, how can i know on which interface is connected my ISDN line??
For outgoing call, i name the channel ZAP/1 in extensions.conf file, but
i dont know if it's correct.
And i always have the message timeout, but no rule 't' in context
What's mean??
There is no extension named t in that context to handle timeouts.
Your dialplan reads:
[PSTN]
exten = 1,1,Dial (IAX2/300,20)
exten = s,2,Voicemail, u300)
So no timeout action is specified. Ignore it if you don't just want to
have the call disconnected on timeout without taking any other action.
I'm not sure if the space after Dial is legal. I figure it may be the
source to your problem. Do you get an error in the CLI when reloading?
Before reloading:
set verbose 1
to see only the relevant warnings.
I have the same message!
Do you know how i can stop messages from qozap (they fill the screen
either asterisk is down!!!)
Best regards,
--
Olivier Saulnier
STEGANUX
1er étage Diamecans
Bel Air
03410 St Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--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] TDM400P static on call
Derek Lee-Wo wrote: I just got a TDM400P with 2 FXO cards. I got it all configured and I can place and receive calls. I seem to be getting static on the call, mainly when I speak. E.g, if I call someone, I can hear them just fine, but they would hear static. Not a lot...more like a constant background hissing noise. What can I look at to try and troubleshoot what is going on, or is this normal? I played with the rx and tx values, but it didn't seem to make much difference. I've had the same problem plus low volume audio when the two lines are bridged.. Adjusting rxgain/txgain does make it louder but also increases the noise. In your case make sure you're not sharing irq's. cat /proc/interrupts Also turn off printer/usb/serial (if you're not using them) in the bios. JD ___ --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] Compiling zaptel on FC5
Are there any generic install guidelines for compiling the Zaptel drivers on FC5? This is my first install of Asterisk (and my first FC5 system) and I'm having a great deal of trouble getting it to cooperate. make clean and make are definitely not playing nice, telling me that "You don't appear to have the kernel sources installed" when I'm pretty sure that I do. Any pointers? thanks, JJ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream GXV-3000
I think here are 2 mixed subject One Substitution of the GXP 3000 Video Phone Phone with a great speaker phone. For the second Subject a think Polycom are the greatest. On 6/13/06, Steve Underwood [EMAIL PROTECTED] wrote: Mike Fedyk wrote: Or any polycom phone that has speakerphone like the IP501 and IP430. Time Bandit wrote: Can you, or anyone else comment on the speakerphone ability of the GVX-3000 ? We run the GXP-2000's and for the most part are happy with them, but for upper management we're looking at phones with better speakerphone. These would be ideal if the speakerphone isn't as terrible as the GXP-2000. I never tried the GVX-3000, but I can recommend with confidence a Cisco 7940 or 7960 for the quality of the speakerphone. And how good do you find the video on the Cisco 7940, 7960 and PolycomIP501, IP430? Personally, I've never seen them do video, but if you aresuggesting them as a substitute for a GVX-3000 I guess I must be wrong. :-) Steve___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as Wholesale
a2billing does both prepaid postpaid accounts. Each account hasa calling card but you can add a sip or iax friend to that card. So, you give a card $20, and that card also has a sip user attached... the sip user will also have $20. RTFM a little closer ;-) bp On 6/12/06, Daniel Salama [EMAIL PROTECTED] wrote: This seems to be only for prepaid calling cards. Is there something that can also handle prepaid and postpaid multi-tenant SIP phones? - Daniel On Jun 12, 2006, at 5:00 PM, William Piper wrote: www.asterisk2billing.org On 6/12/06, Wasif [EMAIL PROTECTED] wrote: Hi,I need to use Asterisk as a switch which can handle wholesale traffic withbilling. Please advice me how I can I implement this. Thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling zaptel on FC5
J.J. Feminella wrote: Are there any generic install guidelines for compiling the Zaptel drivers on FC5? This is my first install of Asterisk (and my first FC5 system) and I'm having a great deal of trouble getting it to cooperate. make clean and make are definitely not playing nice, telling me that You don't appear to have the kernel sources installed when I'm pretty sure that I do. Any pointers? Make sure you have the kernel-devel package installed for the currently running kernel. make ; make install in the libpri directory (if you need PRI support) make linux26; make install in the zaptel directory make ; make install in the asterisk directory This is the basic procedure that I follow when I install/upgrade to a newer version of Zaptel/libpri/Asterisk. Steve ___ --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] Which simple billing application
Hello, I look at voip-info for a simple billing application . I wish to calculate price to pay according to the datas stored in cdr table (unixodbc/mysql). what do you advise me ? Harry __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: TTS from MySQL
Both Festival and Cepstral app commands take a string but you could use
external programming to pass a file or in this modified example from The
Future of Telephony use a filename.
Lines indented are all on one line.
(copy text to file)
echo All circuits are busy. Please try your call later.
/var/lib/asterisk/sounds/text/all-busy
(setup extenxion to test)
exten = 528,1,Answer()
exten = 528,2,Macro(saytext,all-busy)
exten = 528,3,Hangup()
(macro calls Cepstral swift. Could be festival)
[macro-saytext]
exten = s,1,Setvar(text_filename=/var/lib/asterisk/sounds/text/${ARG1})
exten = s,2,System(/usr/local/bin/swift -p 'audio/sampling-rate=8000' -n
Diane -m text -f ${text_filename} -o /tmp/swift.wav)
exten = s,3,Playback(/tmp/swift)
exten = s,4,System(rm /tmp/swift.wav)
The only thing annoying is that if the string is very long, a page of text
for instance, there is a considerable delay while it makes the wav file.
For short, one line sentences, this is not a problem.
Doug
On Mon, 12 Jun 2006, Walid Azab wrote:
Hi all,
I need to simply use Asterisk to receive incoming calls in an IVR manner. It
should authenticate users and read data from MySQL table that match their ID
through Text-to-speech. I already have Asterisk 2.6 ([EMAIL PROTECTED]). I
understand that I need to use Festival and AGI but do not know what to do
exactly. Any help is appreciated.
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
Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety. -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Vs SIP cpu load
Hello Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads? Also, does Asterisk support and use multiprocessor architectures, such as Xeon? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which simple billing application
Hi,Well, I'm working with a2billing http://www.asterisk2billing.org/, without problems.RegardsOn 6/13/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello,I look at voip-info for a simple billing application .I wish to calculate price to pay according to thedatas stored in cdr table (unixodbc/mysql).what do you advise me ?Harry __Do You Yahoo!?En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Vs SIP cpu load
dual support yes.. however i read a few articles on the fuct that single with double the ram is better..something about the bus or sshare between both processors i think.i would go AMD opteron, but that me. or sunOn 6/13/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: HelloIs it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads?Also, does Asterisk support and use multiprocessor architectures, such as Xeon? RegardsJon--No virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Managerhttp://www.theclubvoip.comMaking it happen 1.888.470.7253 ___ --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] echo sidetone grandstream and tdm400p
Hi all, thanks to the all of you. This list is very interesting also for a newby like me. My problem: I just setup my first full working asterisk installation with this config: 1. n.1 GXP-2000 2. n.4 Budgetone 102 3. n.1 TDM400p (3 FXS, 1 FXO) Everything seems to work fine, but the sidetone... it's really annoying! We can hear the sidetone only when we call to the outside (PSTN), it doesn't matter if we call a local, a mobile or a longdistance call. Only we hear the echo, not the called party. We do not ear any echo in internal call to each other extensions. I tryed every possible setting of the echotraining, of the rx and of the tx gain, but with no success. Any idea or help? Thank you in advance, Marco __ Dott. Ing. Marco Sajeva Visioni - we network http://www.visioni.info ___ --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
I use IAX2 quite a bit and I haven't really noticed any difference between
IAX2 and SIP. CPU usage in Asterisk is aggravated by transcoding, changing
one audio format to another, and SIP or IAX2 is simply the protocol used to
carry the audio. Any function of Asterisk will be affected by high system
load; if you have a loadaverage of 3, for example, your box is in trouble
regardless of the protocol used.
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
dispatching *internal* processes to different CPU's, instead, *external*
processes such as AGI's are balanced out and dispatched automatically to
different CPU's - but this is a kernel thing.
It's generally well-known that a fake SMP machine such as a HyperThreading
CPU affects Asterisk negatively, and best practice is to disable
HyperThreading. However, real SMP machines have no trouble (I use a 4 way
Xeon). It's possible to pin a process to a specific CPU, and in fact, I do
this to force Asterisk to it's own CPU, and pin all other processes to a
specific CPU that Asterisk does *not* use:
setasteriskaffinity.sh:
#!/bin/bash
ASTERISKPID=`ps -A | grep -a -A0 asterisk`
taskset 0x0003 -p ${ASTERISKPID:0:5}
This pins Asterisk to CPU # 4 on a 4 way system. Repeat for all other
processes, to different CPU's with the affinity mask:
0x = CPU 1
0x0001 = CPU 2
0x0002 = CPU 3
0x0003 = CPU 4
-Original Message-
From: Jon Schøpzinsky [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX2 Vs SIP cpu load
Hello
Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that
IAX2 is much more sensitive against high CPU loads?
Also, does Asterisk support and use multiprocessor architectures, such as
Xeon?
Regards
Jon
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Eyebeam chat function
Unless it's changed recently, Asterik doesn't support the SIP 'MESSAGE' command. Doug. -Original Message- From: Attilla De Groot [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 2:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Eyebeam chat function Hi all, Eyebeam has a sip-chat function and it would be nice if I would be able to use it. But the problem is that I can't really find information about it. I can just try to send a message and on the Asterisk console a message like this appears: Jun 13 10:05:25 WARNING[6512]: chan_sip.c:7281 receive_message: Received message to sip:[EMAIL PROTECTED] from Bla Sheepsip:[EMAIL PROTECTED];tag=1d072048, dropped it... Content-Type:text/plain Message: ? Can anyone tell me more about this or give me a link with some information about it ? Regards, Attilla de GrootÎ ___ --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 sidetone grandstream and tdm400p
Turn down your microphone TX gains on the phones. On my TDM400 with Vista 350's I had to crank the mic value way down. This is not specific to FXS phones, on my Snom 200's sidetone is so bad, that an appropriate setting for mic gain is '2' (out of 8) hth -Original Message- From: Marco Sajeva [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 8:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] echo sidetone grandstream and tdm400p Hi all, thanks to the all of you. This list is very interesting also for a newby like me. My problem: I just setup my first full working asterisk installation with this config: 1. n.1 GXP-2000 2. n.4 Budgetone 102 3. n.1 TDM400p (3 FXS, 1 FXO) Everything seems to work fine, but the sidetone... it's really annoying! We can hear the sidetone only when we call to the outside (PSTN), it doesn't matter if we call a local, a mobile or a longdistance call. Only we hear the echo, not the called party. We do not ear any echo in internal call to each other extensions. I tryed every possible setting of the echotraining, of the rx and of the tx gain, but with no success. Any idea or help? Thank you in advance, Marco __ Dott. Ing. Marco Sajeva Visioni - we network http://www.visioni.info ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with VoicemailMain
Hi, I'm running SER with Asterisk, and I've configured VoicemailMain like this: exten = 201,1,VoicemailMain(@default) exten = 201,2,Hangup() Although, after any user enter his voicemailmain mailbox, when the phone is hung up, the call still continues running in Asterisk, because I can see it in the debug output of the Asterisk CLI. The call only stops if before hung up, I press #. What is causing this? Any ideas? Regards, Ricardo. ___ --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] Compiling zaptel on FC5
Try "rpm -qa kernel-devel" or "rpm -qa kernel-smp-devel" to verify they're installed. If not do a "yum install kernel-devel or kernel-smp-devel" depending on which you have. Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J.J. FeminellaSent: June 13, 2006 9:51 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Compiling zaptel on FC5 Are there any generic install guidelines for compiling the Zaptel drivers on FC5? This is my first install of Asterisk (and my first FC5 system) and I'm having a great deal of trouble getting it to cooperate. make clean and make are definitely not playing nice, telling me that "You don't appear to have the kernel sources installed" when I'm pretty sure that I do. Any pointers? thanks, JJ ___ --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] WG: Dialplan problem with Digium tdm04p card
hi all, i'am new in the asterisk business and i have to solve following experimental problem: ** user 1 (calls number 12345) -pstn line --*FXO 1FXO 2*pstn line --- user 2 (with number ) * *(asterisk calls ) * ASTERISK PBX * * * user 2 (calls number 67890) -pstn line -- *FXO 3 FXO 4*pstn line --- user 4 (with number ) ** does anybody had to solve a similar problem and could show me a dialplan? tks in advanced franky ___ --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?
Well thanks all for your responses. My original intention was to address the mistic know-how about machine calculations, and I still feel the shadows remain. Why? Because to achieve a 24 user PBX-only/One E1, I was going to install a Dual Xeon with 2 GB of RAM and a 3ware card runing RAID-1 with two sata3 disks. Now This thread tells me that my dual core pentium d (a 700$ computer) will do the work. (the other equipment costs about 3500.00$). I do realize that i must minimize transcoding (ulaw all the way) but you're telling me it will work for 24 users (let's say 30 for round numbers) all with SIP phones in an IP network. Below are some comments that i found googling and doing some calculations myself. I do not enforce or deny any of them, please feel free to tell me if Im wrong. (not confirmed)a- A voice channel takes 30mhz (60mhz in duplex mode). So A 2.66 Ghz CPU can sustain about 43 calls (2600mhz/60mhz=43calls), not taking into account other factors that may increase/decrease the number of calls at the same time. b- 24 users talking ulaw (-+ 80kbps per channel) consume 1920kbps and in full duplex they consume 3840kbps (about 3.75 megabits/s). c- To Calculate the bandwidth DDR memory can achieve (example PC4200) ,to get the transfer rate, multiply the width of the module (8 Bytes) by the rated speed of the memory module (in MHz): (8 Bytes) x (533 MHz/second) = 4,264 Mbytes/second or 4.2 Gbytes/second (34gigabits/s), hence the name PC4200 So, will all of this in mind, CPU Dual Core 533FSB, 2.66 Ghz speed DDR533mhz, One gigabyte. (2x512) Two Sata disks (each sata pumps 1.5 gigabits/s) Motherboard Intel 945 at 533FSB Means that the cpu,the ram and the board can achieve (see point b) about 34 gigabits of data transfer, but 24 users only generate 3.75 megabits. So this is more than covered. However if we take into account the lowest performing component on this system (the sata disks) we go down to 1.5gbits/s which still seems to be enough. Please please correct me if im wrong (or crazy) Thanks, References: http://en.wikipedia.org/wiki/Front_side_bus (bus bandwidth table) http://www.acme.com/build_a_pc/bandwidth.html http://www.lostcircuits.com/memory/ddrii/ http://bvio.ngic.re.kr/Bvio/index.php?title=Front_side_bus On 6/13/06, Mike Fedyk [EMAIL PROTECTED] wrote: 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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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
When i did this test ages ago, i found out that iax was worse than sip,
but sip was worse than trunked iax.
Joachim
olin Anderson wrote:
I use IAX2 quite a bit and I haven't really noticed any difference between
IAX2 and SIP. CPU usage in Asterisk is aggravated by transcoding, changing
one audio format to another, and SIP or IAX2 is simply the protocol used to
carry the audio. Any function of Asterisk will be affected by high system
load; if you have a loadaverage of 3, for example, your box is in trouble
regardless of the protocol used.
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
dispatching *internal* processes to different CPU's, instead, *external*
processes such as AGI's are balanced out and dispatched automatically to
different CPU's - but this is a kernel thing.
It's generally well-known that a fake SMP machine such as a HyperThreading
CPU affects Asterisk negatively, and best practice is to disable
HyperThreading. However, real SMP machines have no trouble (I use a 4 way
Xeon). It's possible to pin a process to a specific CPU, and in fact, I do
this to force Asterisk to it's own CPU, and pin all other processes to a
specific CPU that Asterisk does *not* use:
setasteriskaffinity.sh:
#!/bin/bash
ASTERISKPID=`ps -A | grep -a -A0 asterisk`
taskset 0x0003 -p ${ASTERISKPID:0:5}
This pins Asterisk to CPU # 4 on a 4 way system. Repeat for all other
processes, to different CPU's with the affinity mask:
0x = CPU 1
0x0001 = CPU 2
0x0002 = CPU 3
0x0003 = CPU 4
-Original Message-
From: Jon Schøpzinsky [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX2 Vs SIP cpu load
Hello
Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that
IAX2 is much more sensitive against high CPU loads?
Also, does Asterisk support and use multiprocessor architectures, such as
Xeon?
Regards
Jon
___
--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?
Maybe you are over thinking it. I have 200 users on a quad Xeon 700 with 2 PRI's and we regularly have 40-60 channels up, no problem (believe me, if there was a problem I'd have 200 guys freaking on my head). I rarely see 30% single-CPU usage, and that's only when Sendmail is invoked to send out a voicemail. But yes, transcoding and reasonable echocancel values is key. If you are connecting to the PSTN, ulaw all the way. If you are connecting to a provider, use the codec of your choice as long as your provider supports it, and make sure every phone and endpoint is set to use the same codec. I also have 30 IAX remote sites that support from 1 to 5 users, on P-II 233's. I use them because they are bulletproof and they are so cheap if something gets hosed we just throw it away and put in another one. Again, no problem Maybe try your cheapo machine and if it doesn't work try a better box. You already have the cheap machine, and the card will remain the same regardless of what box you use. -Original Message- From: Erick Perez [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 9:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can this config sustain 30 users? Well thanks all for your responses. My original intention was to address the mistic know-how about machine calculations, and I still feel the shadows remain. Why? Because to achieve a 24 user PBX-only/One E1, I was going to install a Dual Xeon with 2 GB of RAM and a 3ware card runing RAID-1 with two sata3 disks. Now This thread tells me that my dual core pentium d (a 700$ computer) will do the work. (the other equipment costs about 3500.00$). I do realize that i must minimize transcoding (ulaw all the way) but you're telling me it will work for 24 users (let's say 30 for round numbers) all with SIP phones in an IP network. Below are some comments that i found googling and doing some calculations myself. I do not enforce or deny any of them, please feel free to tell me if Im wrong. (not confirmed)a- A voice channel takes 30mhz (60mhz in duplex mode). So A 2.66 Ghz CPU can sustain about 43 calls (2600mhz/60mhz=43calls), not taking into account other factors that may increase/decrease the number of calls at the same time. b- 24 users talking ulaw (-+ 80kbps per channel) consume 1920kbps and in full duplex they consume 3840kbps (about 3.75 megabits/s). c- To Calculate the bandwidth DDR memory can achieve (example PC4200) ,to get the transfer rate, multiply the width of the module (8 Bytes) by the rated speed of the memory module (in MHz): (8 Bytes) x (533 MHz/second) = 4,264 Mbytes/second or 4.2 Gbytes/second (34gigabits/s), hence the name PC4200 So, will all of this in mind, CPU Dual Core 533FSB, 2.66 Ghz speed DDR533mhz, One gigabyte. (2x512) Two Sata disks (each sata pumps 1.5 gigabits/s) Motherboard Intel 945 at 533FSB Means that the cpu,the ram and the board can achieve (see point b) about 34 gigabits of data transfer, but 24 users only generate 3.75 megabits. So this is more than covered. However if we take into account the lowest performing component on this system (the sata disks) we go down to 1.5gbits/s which still seems to be enough. Please please correct me if im wrong (or crazy) Thanks, References: http://en.wikipedia.org/wiki/Front_side_bus (bus bandwidth table) http://www.acme.com/build_a_pc/bandwidth.html http://www.lostcircuits.com/memory/ddrii/ http://bvio.ngic.re.kr/Bvio/index.php?title=Front_side_bus On 6/13/06, Mike Fedyk [EMAIL PROTECTED] wrote: 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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Follow Me
Is there a way to patch an existing Asterisk 1.2.5 version with the follow me application? -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Saturday, February 25, 2006 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Follow Me On 2/25/06, Nilesh Londhe [EMAIL PROTECTED] wrote: I could not find followme app listed when I tried show applications on the CLI. Is this app patch incorporated into asterisk 1.24 release tree? If not, what are the plans for the future? On 2/24/06, Dinesh Nair [EMAIL PROTECTED] wrote: On 02/23/06 23:08 Darrick Hartman said the following: True, but why not accept the app? It sure makes the dial plan alot nothing wrong with that, i wasnt suggesting rejecting the application or anything. just pointing out that scripting it within the dialplan makes it more flexible for more people, especially those who cant code in C to change how it behaves. It's not part of the main tree yet. I don't really know whether or not it will make it into v1.4. I hope so, but it's not up to me. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
taskset does not seem to exist on redhad 9 nor freebsd..;)On 6/13/06, Zoa [EMAIL PROTECTED] wrote:
When i did this test ages ago, i found out that iax was worse than sip,but sip was worse than trunked iax.
Joachimolin Anderson wrote: I use IAX2 quite a bit and I haven't really noticed any difference between IAX2 and SIP. CPU usage in Asterisk is aggravated by transcoding, changing one audio format to another, and SIP or IAX2 is simply the protocol used to
carry the audio. Any function of Asterisk will be affected by high system load; if you have a loadaverage of 3, for example, your box is in trouble regardless of the protocol used. 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 dispatching *internal* processes to different CPU's, instead, *external* processes such as AGI's are balanced out and dispatched automatically to
different CPU's - but this is a kernel thing. It's generally well-known that a fake SMP machine such as a HyperThreading CPU affects Asterisk negatively, and best practice is to disable
HyperThreading. However, real SMP machines have no trouble (I use a 4 way Xeon). It's possible to pin a process to a specific CPU, and in fact, I do this to force Asterisk to it's own CPU, and pin all other processes to a
specific CPU that Asterisk does *not* use: setasteriskaffinity.sh: #!/bin/bash ASTERISKPID=`ps -A | grep -a -A0 asterisk` taskset 0x0003 -p${ASTERISKPID:0:5}
This pins Asterisk to CPU # 4 on a 4 way system. Repeat for all other processes, to different CPU's with the affinity mask: 0x = CPU 1 0x0001 = CPU 2
0x0002 = CPU 3 0x0003 = CPU 4 -Original Message- From: Jon Schøpzinsky [mailto:[EMAIL PROTECTED]] Sent: Tuesday, June 13, 2006 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAX2 Vs SIP cpu load Hello Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that
IAX2 is much more sensitive against high CPU loads? Also, does Asterisk support and use multiprocessor architectures, such as Xeon? Regards Jon
___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Manager
http://www.theclubvoip.comMaking it happen1.888.470.7253
___
--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
2002
called. They want their operating system back. :- )
-Original Message-From: Mike Lynchfield
[mailto:[EMAIL PROTECTED]Sent: Tuesday, June 13, 2006 9:42
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] IAX2 Vs SIP cpu
loadtaskset does not seem to exist on redhad 9 nor
freebsd..;)
On 6/13/06, Zoa
[EMAIL PROTECTED] wrote:
When
i did this test ages ago, i found out that iax was worse than sip,but
sip was worse than trunked iax. Joachimolin Anderson
wrote: I use IAX2 quite a bit and I haven't really noticed any
difference between IAX2 and SIP. CPU usage in Asterisk is aggravated
by transcoding, changing one audio format to another, and SIP or
IAX2 is simply the protocol used to carry the audio. Any function
of Asterisk will be affected by high system load; if you have a
loadaverage of 3, for example, your box is in trouble regardless of
the protocol used. 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 dispatching *internal* processes
to different CPU's, instead, *external* processes such as AGI's are
balanced out and dispatched automatically to different CPU's - but
this is a kernel thing. It's generally well-known that a
"fake" SMP machine such as a HyperThreading CPU affects Asterisk
negatively, and best practice is to disable HyperThreading.
However, "real" SMP machines have no trouble (I use a 4 way Xeon).
It's possible to "pin" a process to a specific CPU, and in fact, I
do this to force Asterisk to it's own CPU, and pin all other
processes to a specific CPU that Asterisk does *not*
use: setasteriskaffinity.sh:
#!/bin/bash ASTERISKPID=`ps -A | grep -a -A0 "asterisk"`
taskset 0x0003 -p${ASTERISKPID:0:5} This
"pins" Asterisk to CPU # 4 on a 4 way system. Repeat for all other
processes, to different CPU's with the affinity mask:
0x = CPU 1 0x0001 = CPU 2 0x0002 = CPU
3 0x0003 = CPU 4 -Original
Message- From: Jon Schøpzinsky [mailto:[EMAIL PROTECTED]] Sent: Tuesday, June
13, 2006 8:14 AM To: Asterisk Users Mailing List - Non-Commercial
Discussion Subject: [Asterisk-Users] IAX2 Vs SIP cpu
load Hello Is it correct that IAX2
uses more CPU, than SIP? Also, can it be true that IAX2 is much
more sensitive against high CPU loads? Also, does Asterisk support
and use multiprocessor architectures, such as
Xeon? Regards Jon
___--Bandwidth
and Colocation provided by Easynews.com
--Asterisk-Users mailing listTo UNSUBSCRIBE or update options
visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Managerhttp://www.theclubvoip.comMaking it
happen1.888.470.7253
___
--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] Festival RPM?
Hi, is there a RHEL4 RPM for the Festival text-to-speech system? Thanks -- Domenico Viggiani ___ --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 Follow Me
On 6/13/06, Kevin Kiely [EMAIL PROTECTED] wrote: Is there a way to patch an existing Asterisk 1.2.5 version with the follow me application? Not at this time, no. There are a number of API calls in the application that are specific to the new version of Asterisk and will not port back to the 1.2.X version without some work. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival RPM?
um, yum install festival worked for me. -Original Message- From: Mimmus [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 9:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Festival RPM? Hi, is there a RHEL4 RPM for the Festival text-to-speech system? Thanks -- Domenico Viggiani ___ --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
Mike Lynchfield wrote: taskset does not seem to exist on redhad 9 nor freebsd.. ;) On Fedora Core 4, it is provided by the schedutils RPM. Steve ___ --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
Go to:
http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
and search for affinity, iirc i explain there how to do it with echo
instead of tasksel.
Zoa
Colin Anderson wrote:
2002 called. They want their operating system back. :- )
-Original Message-
*From:* Mike Lynchfield [mailto:[EMAIL PROTECTED]
*Sent:* Tuesday, June 13, 2006 9:42 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [Asterisk-Users] IAX2 Vs SIP cpu load
taskset does not seem to exist on redhad 9 nor freebsd..
;)
On 6/13/06, *Zoa* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
When i did this test ages ago, i found out that iax was worse
than sip,
but sip was worse than trunked iax.
Joachim
olin Anderson wrote:
I use IAX2 quite a bit and I haven't really noticed any
difference between
IAX2 and SIP. CPU usage in Asterisk is aggravated by
transcoding, changing
one audio format to another, and SIP or IAX2 is simply the
protocol used to
carry the audio. Any function of Asterisk will be affected
by high system
load; if you have a loadaverage of 3, for example, your box
is in trouble
regardless of the protocol used.
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
dispatching *internal* processes to different CPU's,
instead, *external*
processes such as AGI's are balanced out and dispatched
automatically to
different CPU's - but this is a kernel thing.
It's generally well-known that a fake SMP machine such as
a HyperThreading
CPU affects Asterisk negatively, and best practice is to
disable
HyperThreading. However, real SMP machines have no trouble
(I use a 4 way
Xeon). It's possible to pin a process to a specific CPU,
and in fact, I do
this to force Asterisk to it's own CPU, and pin all other
processes to a
specific CPU that Asterisk does *not* use:
setasteriskaffinity.sh:
#!/bin/bash
ASTERISKPID=`ps -A | grep -a -A0 asterisk`
taskset 0x0003 -p ${ASTERISKPID:0:5}
This pins Asterisk to CPU # 4 on a 4 way system. Repeat
for all other
processes, to different CPU's with the affinity mask:
0x = CPU 1
0x0001 = CPU 2
0x0002 = CPU 3
0x0003 = CPU 4
-Original Message-
From: Jon Schøpzinsky [mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]]
Sent: Tuesday, June 13, 2006 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX2 Vs SIP cpu load
Hello
Is it correct that IAX2 uses more CPU, than SIP? Also, can
it be true that
IAX2 is much more sensitive against high CPU loads?
Also, does Asterisk support and use multiprocessor
architectures, such as
Xeon?
Regards
Jon
___
--Bandwidth and Colocation provided by Easynews.com
http://Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike
Sales Manager
http://www.theclubvoip.com
Making it happen
1.888.470.7253
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sound quality problem on mISDN
Hi I've problem with incoming call quality to GSM gateway connected to beronet card (BN8S0), - [ GSM Gateway ] --- [ BN8S0 ] asterisk Port connected to GSM gatway is in TE mode , gateway is in NT mode , When I dialin to cellphone numer , call goes to 'from-eragsm' context, to Echo application. [from-eragsm] exten = 700,1,Goto(600,1) exten = 600,1,Answer() exten = 600,2,Playback(demo-echotest) exten = 600,n,Echo exten = 600,n,Playback(demo-echodone) misdn.conf: [eragsm-gw] ports=1ptp context=from-eragsm nationalprefix=0 internationalprefix=00 echocancel=yes echocancelwhenbridged=no dialplan=2 msns=600,700 Everything is good besides call quality, sound is choppy, with lot of noises, when I tell one , two , three ... test , I hear only three, sometimes more , I've already tried to increase rxgain/txgain for this channel , but It didn't help much. Outgoing call quality is rather normal. TIA for any help with this. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --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
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 ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hard drive write cache
The cold hard truth is that if Asterisk cannot achieve 99.999% uptime without becoming much more expensive that a traditional PBX then it is not a viable alternative. Even elcheapo Key systems are rated for five nines. That is what the telco world requires unless your just using Asterisk in your basement as a hobby or as a one man company. Redundant Servers is moving into the realm of non-competitive with Traditional PBX IMHO. I don't care about corruption of the CDR or any of the logging/database information. All I care about is the ability make phone calls after power failure. That IS the MAIN function of a PBX. Not call centers, databases, CDR, click 2 call, and all the other bells and whistles. -Original Message- From: Boris Bakchiev [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 2:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Hard drive write cache These days you don't have to worry much about your write cache unless you're running application where once single byte changed will affect whole file. Look at it this way, the only corruption will occur is whatever the files were open by asterisk at the time of the crash. And only up to the point where the file was last open. As far as I know asterisk does not keep cdr or log files open so you would loose only the data that was written at the time of the power failure. Any journaling file system (ext3, resierfs, xfs, etc) will easily handle any power failure event. Your files will not be corrupt but could miss some of the data. At the most you will loose 10-50 cdr entries written to you log files. If you post CDR to a remote SQL database then you asterisk install and linux is more or less static and will not be affected by the power failure. What you need to do is minimise the writes to hard disk's: 1 - Send syslog to remote server and do not do ANY syslogs Or keep the circular buffer in memory if you have plenty of it. 2 - Send CDR's to SQL server (or log to ramdisk and send to remote server every few minutes via SSH) 3 - Do not record any calls (or do that somewhere else) 4 - Stop any services that write/read data on regular intervals. If you have no writes you have nothing to worry about during power failure and journaling file system will take care of the rest. Keep your partition size really small so that fsck will not take much time. You have to be realistic, you cannot achieve 99.999% uptime. That's 5 minutes per year downtime. You will have more or less 100% until your first hardware failure. Even if you have all the hardware components pre-purchased it will still take you 2-12 hours to detect, diagnose and fix the fault if you lucky. So your 5 minuets If the business is demanding 99.999% then it should be prepared to invest into the hardware. I would recommend a cluster or even better a fault tolerant server. Those are expensive but you can pretty much rule out the hardware failure and swap all of the failed components while the system is running (cpu, memory, hdd, etc). Look at Stratus or NEC FT servers if you need hardware redundancy. They're expensive but will give you the hardware reliability you need. Or get a traditional PABX :) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of shadowym Sent: Tuesday, 13 June 2006 10:34 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hard drive write cache I am looking at ways to harden my asterisk install to prevent computer related issues from happening. I am concerned about about disk write cache. That seems to be a major source of hard drive corruption on power failure. Hard Drive corruption is simply unacceptable for the 99.999% uptime requirements of my Asterisk install that needs to be as reliable as a proprietary PBX. Of course I will be using redundant power supplies, raid 1 and use a UPS. None of those things mean much if the power cords accidentally get pulled from the back of the server. Unlikely as it may be I have to consider ALL possibilities. So is disabling the write cache a good way to reduce the risk of hard drive corruption for an Asterisk server? I am not too concerned about the reduced performance/lifetime of hardrives with write cache disabled since Asterisk is not a very write intensive environment. Even with lot's of voicemail going on. Any other recommendations/links for increasing the reliability of Asterisk servers? ___ --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
Re: [Asterisk-Users] Can this config sustain 30 users?
Hi Erick - Now This thread tells me that my dual core pentium d (a 700$ computer) will do the work. (the other equipment costs about 3500.00$). I do realize that i must minimize transcoding (ulaw all the way) but you're telling me it will work for 24 users (let's say 30 for round numbers) all with SIP phones in an IP network. I do more with less hardware without any problems. In one of our offices I have a single P4 2.8Ghz with 1GB Ram and two SATA RAID 1 drives. This machine currently serves about 35 sip phones (about 25 actively used), has a TE410P with one PRI, a TDM22 handling 2 backup POTS lines on the FXO's and fax machines on the FXS's, and connects to 6 other offices via IAX that transcode to GSM. It also acts as a DNS resolver, the FTP server for the Polycom SIP phones, and runs NTP. At one point I had it running FOP, which is fairly resource intensive. It could very easily handle much more. Top shows the CPU is between 92% (12 active channels) and 99.8% (2 active channels) idle, and memory usage at about 470MB total, 290MB active. All this is to say that you have plenty of hardware for your current needs and considerable growth. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Repost] Asterisk realtime
Hi folks, I'm really confused, so please help me, or at least give me some pointers to clarify this issue. Can I mix Static and Real realtime? Is there a way to easily switch from one to another, say, for sip.conf? Which are the major benefits of Real realtime? Please help me! Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --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] Queues and macros and agents
- Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Now, I want to be able to use a device, rather than agents. So I can
use
addQueueMember and add my SIP device. However, I still want to do a
couple of things before the device is called.
This is what the Local channel (chan_local) is for.
If your SIP device is called myfancyphone, then instead of adding
SIP/myfancyphone to the queue using AddQueueMember, add (instead) Local/[EMAIL
PROTECTED], and then in your dialplan:
[members]
exten = myfancyphone,1,...
exten = myfancyphone,n,...
exten = myfancyphone,n,Dial(SIP/${EXTEN})
--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Bounty Doubling program
TRX Teleocmmunications the VoIP provider that pays you would like to assist those that make asterisk better. To that end we are setting up a program where the community itself can help double the bounty for all of the outstanding code that is wanted but not yet present. TRX will match any bounty paid on any new code that gets put into tree in response to a bounty listed at http://www.voip-info.org/wiki-Asterisk +bounty. There are some rules for this doubling program which are available at http://www.trxtel.com/index.php?page=Asterisk_Bounty in short we will donate a small bit of money for each minute that each person is on the phone to a tollfree north american number (we will be having the same program for inbound DIDs soon as well). When a bounty is claimed, we will match what is paid. Asterisk gets more features, developers get more incentive, and the community as a whole can help make that happen. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com The VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --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?
Don't forget to be sure your power supplies are reliable, and if necessary redundant. On 6/13/06, Colin Anderson [EMAIL PROTECTED] wrote: Maybe you are over thinking it. I have 200 users on a quad Xeon 700 with 2PRI's and we regularly have 40-60 channels up, no problem (believe me, if there was a problem I'd have 200 guys freaking on my head). I rarely see 30% single-CPU usage, and that's only when Sendmail is invoked to send out avoicemail.But yes, transcoding and reasonable echocancel values is key. If you are connecting to the PSTN, ulaw all the way. If you are connecting to aprovider, use the codec of your choice as long as your provider supports it,and make sure every phone and endpoint is set to use the same codec. I also have 30 IAX remote sites that support from 1 to 5 users, on P-II233's. I use them because they are bulletproof and they are so cheap ifsomething gets hosed we just throw it away and put in another one. Again, no problemMaybe try your cheapo machine and if it doesn't work try a better box. Youalready have the cheap machine, and the card will remain the same regardlessof what box you use.-Original Message- From: Erick Perez [mailto:[EMAIL PROTECTED]]Sent: Tuesday, June 13, 2006 9:15 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Can this config sustain 30 users? Well thanks all for your responses. My original intention was toaddress the mistic know-how about machine calculations, and I stillfeel the shadows remain.Why? Because to achieve a 24 user PBX-only/One E1, I was going to install a Dual Xeon with 2 GB of RAM and a 3ware card runing RAID-1with two sata3 disks.Now This thread tells me that my dual core pentium d (a 700$ computer)will do the work. (the other equipment costs about 3500.00$). I dorealize that i must minimize transcoding (ulaw all the way) but you'retelling me it will work for 24 users (let's say 30 for round numbers)all with SIP phones in an IP network.Below are some comments that i found googling and doing some calculations myself. I do not enforce or deny any of them, please feelfree to tell me if Im wrong.(not confirmed)a- A voice channel takes 30mhz (60mhz in duplex mode).So A 2.66 Ghz CPU can sustain about 43 calls (2600mhz/60mhz=43calls), not taking into account other factors that may increase/decrease thenumber of calls at the same time.b- 24 users talking ulaw (-+ 80kbps per channel) consume 1920kbps andin full duplex they consume 3840kbps (about 3.75 megabits/s).c- To Calculate the bandwidth DDR memory can achieve (example PC4200),to get the transfer rate, multiply the width of the module (8 Bytes)by the rated speed of the memory module (in MHz): (8 Bytes) x (533 MHz/second) = 4,264 Mbytes/second or 4.2 Gbytes/second (34gigabits/s),hence the name PC4200So, will all of this in mind,CPU Dual Core 533FSB, 2.66 Ghz speedDDR533mhz, One gigabyte. (2x512)Two Sata disks (each sata pumps 1.5 gigabits/s)Motherboard Intel 945 at 533FSBMeans that the cpu,the ram and the board can achieve (see point b)about 34 gigabits of data transfer, but 24 users only generate 3.75megabits. So this is more than covered. However if we take into account the lowest performing component onthis system (the sata disks) we go down to 1.5gbits/s which stillseems to be enough.Please please correct me if im wrong (or crazy) Thanks,References:http://en.wikipedia.org/wiki/Front_side_bus (bus bandwidth table) http://www.acme.com/build_a_pc/bandwidth.htmlhttp://www.lostcircuits.com/memory/ddrii/ http://bvio.ngic.re.kr/Bvio/index.php?title=Front_side_busOn 6/13/06, Mike Fedyk [EMAIL PROTECTED] wrote: 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--Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de PanamaCel Panama. +(507) 6694-4780___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update
[Asterisk-Users] [Repost] Asterisk realtime
Hi Andrea, yes you can, static realtime helps you out with replacing config files one by another. The real realtime you mean is for the dynamic backend, which gives you the ability to be reread with every call, so asterisk doesnt even need a reload. So the main difference between static and real is simply the static part only gets loaded at startup/reload. The real realtime asterisk looks up every time something happens e.g. a call is being made. Main configuration for static and real is made in /etc/asterisk/extconfig.conf you should although never mix up both in one particular type. What I mean is if you add IAX realtime support with something like iaxpeers = mysql,asterisk,iax_peers iaxusers = mysql,asterisk,iax_users you shouldn't make an additional static like here: iax.conf = mysql,asterisk Here you will find all information you will need, to setup the static and real realtime: http://www.voip-info.org/wiki-Asterisk+RealTime Hope this helped -- Mit freundlichen Grüßen, Best regards, Michael E. Kromer IT Specialist Linux Professional Institute Certified (LPIC) +--+ | CC Computer Consultants GmbH| | ENTERPRISE. IT. BUSINESS. | |==| | AMD Solution Provider| | Sun Microsystems Partner Associate | | Citrix Access Alliance Partner | +--+ smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Vs SIP cpu load
There is work by the devs to threading in the IAX and SIP channels, I believe. I don't know if it's made it's way back to -HEAD or not, maybe kpf can give a definitive answer. I remember reading something by Mark S earlier this year that he had IAX threading working. -Original Message- From: Dinesh Nair [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 Vs SIP cpu load 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 ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can this config sustain 30 users?
Erick, Please see message: Paul Mahler: Asterisk Scalability at the following link: http://asteriskvoip.blogspot.com/2005_06_01_asteriskvoip_archive.html Much slower machine than yours was involved in tests: 47 Simultaneous VoiceMail messages 333 Simultaneous SIP Calls 122 Pass through calls Slightly less than 47% CPU Utilisation I personally think from my experience that Asterisk with wholesale usage is a CPU hog, other components are not that important (depends on usage patterns of course, but we are talking regular office PBX usage). Andrei Erick Perez wrote: Well thanks all for your responses. My original intention was to address the mistic know-how about machine calculations, and I still feel the shadows remain. Why? Because to achieve a 24 user PBX-only/One E1, I was going to install a Dual Xeon with 2 GB of RAM and a 3ware card runing RAID-1 with two sata3 disks. Now This thread tells me that my dual core pentium d (a 700$ computer) will do the work. (the other equipment costs about 3500.00$). I do realize that i must minimize transcoding (ulaw all the way) but you're telling me it will work for 24 users (let's say 30 for round numbers) all with SIP phones in an IP network. Below are some comments that i found googling and doing some calculations myself. I do not enforce or deny any of them, please feel free to tell me if Im wrong. (not confirmed)a- A voice channel takes 30mhz (60mhz in duplex mode). So A 2.66 Ghz CPU can sustain about 43 calls (2600mhz/60mhz=43calls), not taking into account other factors that may increase/decrease the number of calls at the same time. b- 24 users talking ulaw (-+ 80kbps per channel) consume 1920kbps and in full duplex they consume 3840kbps (about 3.75 megabits/s). c- To Calculate the bandwidth DDR memory can achieve (example PC4200) ,to get the transfer rate, multiply the width of the module (8 Bytes) by the rated speed of the memory module (in MHz): (8 Bytes) x (533 MHz/second) = 4,264 Mbytes/second or 4.2 Gbytes/second (34gigabits/s), hence the name PC4200 So, will all of this in mind, CPU Dual Core 533FSB, 2.66 Ghz speed DDR533mhz, One gigabyte. (2x512) Two Sata disks (each sata pumps 1.5 gigabits/s) Motherboard Intel 945 at 533FSB Means that the cpu,the ram and the board can achieve (see point b) about 34 gigabits of data transfer, but 24 users only generate 3.75 megabits. So this is more than covered. However if we take into account the lowest performing component on this system (the sata disks) we go down to 1.5gbits/s which still seems to be enough. Please please correct me if im wrong (or crazy) Thanks, References: http://en.wikipedia.org/wiki/Front_side_bus (bus bandwidth table) http://www.acme.com/build_a_pc/bandwidth.html http://www.lostcircuits.com/memory/ddrii/ http://bvio.ngic.re.kr/Bvio/index.php?title=Front_side_bus On 6/13/06, Mike Fedyk [EMAIL PROTECTED] wrote: 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk keeps running after hungup untill I press #
Hi, I'm running SER with Asterisk, and I've configured VoicemailMain like this: exten = 201,1,VoicemailMain(@default) exten = 201,2,Hangup() Although, after any user enter his voicemailmain mailbox, when the phone is hung up, the call still continues running in Asterisk, because I can see it in the debug output of the Asterisk CLI. The call only stops if before hung up, I press #. What is causing this? Any ideas? Regards, Ricardo. ___ --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
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. Regards, Patrick ___ --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: The cold hard truth is that if Asterisk cannot achieve 99.999% uptime without becoming much more expensive that a traditional PBX then it is not a viable alternative. Even elcheapo Key systems are rated for five nines. Even massive redundant public exchanges struggle for 99.999% up time. No key system gives that. Its about 1 hour down in 10 years. That is what the telco world requires unless your just using Asterisk in your basement as a hobby or as a one man company. Redundant Servers is moving into the realm of non-competitive with Traditional PBX IMHO. A traditional PBX of any decent size is highly redundant. More so than you will easily achieve with *. How do you expect * to achieve high reliability without redundant servers. I don't care about corruption of the CDR or any of the logging/database information. All I care about is the ability make phone calls after power failure. That IS the MAIN function of a PBX. Not call centers, databases, CDR, click 2 call, and all the other bells and whistles. Steve ___ --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
