Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
You can find it here: http://www.zoiper.com/downloads/free/linux/zoiper201-linux.tar.gz Note that the linux version does not support TLS and SRTP yet. * Instructions: * 1) Download zoiper201-linux.tar.gz 2) Extract Zoiper. If you don't use a GUI application for archive processing, here is the command line: tar zxf zoiper201-linux.tar.gz ./zoiper 3) Start Zoiper. *ZoIPer depends on ALSA library, so it* **must** *be installed! * Zoa Robert Moskowitz wrote: zoa wrote: Have you tried our Zoiper softphone yet (www.zoiper.com) - new version scheduled for in a couple of days ? If so, can you send me any remarks of list so that we can keep those things in mind for future versions ? Do you know where I can get it as an rpm to install on Centos 5 with Gnome? I do not have the time resources to do compiles. I am really a security protocol researcher and would be very interested in seeing what you have done for SIP TLS and SRTP. But for the later, I am all Linux. The one XP system is a corp box that I cannot add any software too. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Modem bridging on Asterisk (no VoIP involved)
Hi everybody. I know maybe this question has been posted some time ago, but I need your updated opinion on the subject. I'm replacing our old pbx with asterisk. I have two TE207 dual pri (e1) cards on a clustered system (one on each node). I absolutely need to connect 4/5 analog extensions with modems, they're being used for remote assistance on very old systems which cannot be upgraded to native IP links. Is there a good hardware that can bridge the e1 lines on the digium te207 card to my modems? A PCI card? An external box? I don't want to relay modem connections over ip, I just need to bridge them internally on the asterisk server: E1 == TE207 == Asterisk == (some hardware with FXS) == modems TIA for your replies. -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem bridging on Asterisk (no VoIP involved)
Hi Alberto, I think that here you can find useful hw: http://www.patapsco.co.uk/ Marino On Jan 23, 2008 9:39 AM, Alberto Pastore [EMAIL PROTECTED] wrote: Hi everybody. I know maybe this question has been posted some time ago, but I need your updated opinion on the subject. I'm replacing our old pbx with asterisk. I have two TE207 dual pri (e1) cards on a clustered system (one on each node). I absolutely need to connect 4/5 analog extensions with modems, they're being used for remote assistance on very old systems which cannot be upgraded to native IP links. Is there a good hardware that can bridge the e1 lines on the digium te207 card to my modems? A PCI card? An external box? I don't want to relay modem connections over ip, I just need to bridge them internally on the asterisk server: E1 == TE207 == Asterisk == (some hardware with FXS) == modems TIA for your replies. -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime problem host='dynamic' in 1.2.26.1
Hello! We are using the 1.2 branch, and upgraded to 1.2.26.1. We ran into some problems when using realtime for peers. We connect the PBX to a sip peer at an ITSP, and when we try to dial the peer we get: Jan 23 09:02:07 VERBOSE[2236] logger.c: -- Executing Dial(SIP/dev02-08c36f28, SIP/[EMAIL PROTECTED]||W) in new stack Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Everything is fine. Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic' Jan 23 09:02:07 WARNING[2236] chan_sip.c: No such host: 989800-out Jan 23 09:02:07 NOTICE[2236] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Jan 23 09:02:07 VERBOSE[2236] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Jan 23 09:02:07 DEBUG[2236] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. I looked in the archives and found this thread: http://lists.digium.com/pipermail/asterisk-users/2007-December/202616.html Here the same problem is discussed for the 1.4 branch, and the result is that the problem should be fixed. But this is still a problem in 1.2 branch. Will this be corrected in a new release, or is this not considered a security fix and hence ignored? Actually isn't this a fix for a security fix... BR, Torbjörn Abrahamsson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rotating CDR records inside mysql - anyone does it?
Thanks. I have about 1 million records, but my machine is not so good. Its a core2 duo with 2 Gig of RAM. When I do only select it takes a few seconds, but some of my reports require joins, and thats a big problem. Thiago Well, i wouldn't recommend delete, as that would keep mysql very unhappy. you could do RENAME TABLE and CREATE TABLE, or mysqldump and TRUNCATE TABLE, but they have to happen almost instantly (without asterisk trying to do INSERT). I have nearly none experience with transactions, but probably those would be helpful. Btw, you can block access to mysql by firewall (to move existing data) or stop mysql (to physycally copy binary database files) and then take it back up - asterisk will post it's CDRs later when db comes accessible. Btw - how many records do you have that it gets slow? On what machine? I currently have 3 million CDR records in MySQL with well created indexes - and most reports are dynamic. Usually from 0 to 2 seconds, but sometimes up to minute for joins :p. Well, that's 2x Quad core xeons of 3GHz and 8Gb RAM (2 of which are used by MySQL indexes). Asterisk is running on same machine. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskIdeas.org :: Comment on submitted ideas
I can't say that ideas are pouring in to AsteriskIdeas.org, but we still have a few ideas worth a discussion. Check them out today, vote or add a comment: http://www.asteriskideas.org I've got some feedback about the requirement to create an account to add comments or posts, but due to blogging spam I felt it was the only solution. I don't want the site filled with links to other, non- related sites. Add your wishlist, your ideas and see what the community says! See you at asteriskideas.org! /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
Zoiper is pretty impressive, it's a simple, neat little client. The one problem I have with it is the keyboard. I've had problems trying to use the keyboard to send DTMF on the current call. The left hand popout keypad is also a little small for my users' taste. It would be nice to have a keyboard hang-up, something like ESC, ditto for things like cancel buttons around the app. I really like the fact it does both SIP and IAX. Onto sillier issues: the icon is nice, but it would be great to have proper gamma anti-aliasing on the mac one. Just my .02 on the free mac os version, I might have to check out the biz edition too. It's all looking good. Good luck with the next release! Simon Simon Elliston Ball [EMAIL PROTECTED] On 23 Jan 2008, at 08:35, Zoa wrote: You can find it here: http://www.zoiper.com/downloads/free/linux/zoiper201-linux.tar.gz Note that the linux version does not support TLS and SRTP yet. * Instructions: * 1) Download zoiper201-linux.tar.gz 2) Extract Zoiper. If you don't use a GUI application for archive processing, here is the command line: tar zxf zoiper201-linux.tar.gz ./zoiper 3) Start Zoiper. *ZoIPer depends on ALSA library, so it* **must** *be installed! * Zoa Robert Moskowitz wrote: zoa wrote: Have you tried our Zoiper softphone yet (www.zoiper.com) - new version scheduled for in a couple of days ? If so, can you send me any remarks of list so that we can keep those things in mind for future versions ? Do you know where I can get it as an rpm to install on Centos 5 with Gnome? I do not have the time resources to do compiles. I am really a security protocol researcher and would be very interested in seeing what you have done for SIP TLS and SRTP. But for the later, I am all Linux. The one XP system is a corp box that I cannot add any software too. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail check
Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Gilberto Nunes wrote: A Monday 14 January 2008 16:25:15, Steve Johnson escreveu: Yeah! I'm just do this right now! But I want more! How can I create some extension to call to user, and pass the information about new voicemail message? Um, wouldn't they be unavailable if someone just left them a voicemail? He's talking about calling their cellphone or pager, not their desk. On our voicemail system it's called out dial. We use it with our sales people. If a message is left in their voice mail box the phone system calls them to tell them there is a new message, authenticate themselves and then let them listen to the message. It lets them know there is a message more quickly and since most of them have free incoming minutes it's cheaper. It was probably great when the system was new, but now that most people have crackberrys an email notification is probably just as good. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskIdeas.org :: Comment on submitted ideas
Johansson Olle E wrote: I can't say that ideas are pouring in to AsteriskIdeas.org, but we still have a few ideas worth a discussion. I entered one and submitted it, but then it seems it was caught in approval mode and never showed up by the time I gave up looking at the site. Now that you mention it I see it's there. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No more audio with 99777 SVN version in certain case
Hi, we have an Debian Etch 4.0 amd64 server with 2 B410P cards. Asterisk SVN r99777 is installed. We tried with mISDN shipped with Asterisk/Zaptel (make b410p) as well as with the latest version from mISDN.org 1.1.7.2. zaptel, ztdummy and crt-ccitt modules are loaded. Output of /dev/zap is: [EMAIL PROTECTED]:~# ls -al /dev/zap total 0 drwxr-xr-x 2 root root 120 2008-01-23 16:42 . drwxr-xr-x 17 root root 4400 2008-01-23 16:54 .. crw-rw 1 root root 196, 254 2008-01-23 16:42 channel crw-rw 1 root root 196, 0 2008-01-23 16:42 ctl crw-rw 1 root root 196, 255 2008-01-23 16:42 pseudo crw-rw 1 root root 196, 253 2008-01-23 16:42 timer The problem we face is following: - start Asterisk - enter in a meetme conf from a SIP Phone - OK, we have Enter PIN Number and then You're the only person ... and then MOH - now we call from outside through ISDN and want to enter the conf - we hear few words from Enter your PIN number (sometimes all the sentance) and then *no more* audio in all Asterisk, for all devices. Sometimes break is created when ISDN party hangup, which means that all the conference went OK. It's now anymore possible to enter conferences or to call voicemails _from any device_: on CLI we see that everything is OK, but silence. Calling from a device to another, SIP iAX or ISDN, is working. Only solution to get it work again is to restart Asterisk. The problem appears since this week when we updated from an old SVN version (05/2007). Thanks for any help. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk optimalization
hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is major bottleneck? qualify imho not. i'm tried set qualify=no, does not help SIP REGISTER packets? this problem persist if no calls are active after restart cpu usage slowly increase. after a ~hour is about 100% which optimalizations do you recommend for ~1500 peers scenario? (behind nat, reregistrations) --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk scalability
Hello, I wonder how Asterisk scales when we increment the Core's or CPU's of one computer. I see that Asterisk is only one process (I guess that it uses threads). But because Asterisk is only one process, this process is always executed in the same CPU. So we can have a 8 Cores server, with one Core running Asterisk, another Core running operating system stuff/other small daemons and 6 idle cores. Is this correct? Why not? If this is correct, increasing CPU number of Asterisk server box would not increase the performance. I don't see any other process that could use other Cores (like transcoding processes, executing dialplan, etc.) Thank you for your information, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskIdeas.org :: Comment on submitted ideas
23 jan 2008 kl. 17.28 skrev Steve Prior: Johansson Olle E wrote: I can't say that ideas are pouring in to AsteriskIdeas.org, but we still have a few ideas worth a discussion. I entered one and submitted it, but then it seems it was caught in approval mode and never showed up by the time I gave up looking at the site. Now that you mention it I see it's there. Yes, while starting this up I am moderating the posts. If this grows, I might need help from others so moderate, but right now it's no problem more than the delay it causes for you. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
On Wed, 23 Jan 2008, Zoa wrote: You can find it here: http://www.zoiper.com/downloads/free/linux/zoiper201-linux.tar.gz Note that the linux version does not support TLS and SRTP yet. * Instructions: * 1) Download zoiper201-linux.tar.gz 2) Extract Zoiper. If you don't use a GUI application for archive processing, here is the command line: tar zxf zoiper201-linux.tar.gz ./zoiper 3) Start Zoiper. I liked Zoiper when it was idefisk however I'm very irritated that they changed the account limit to 2 in Zoiper after it was seemingly unlimited in idefisk, so guess what I stick with... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
Hello, I wonder how Asterisk scales when we increment the Core's or CPU's of one computer. I see that Asterisk is only one process (I guess that it uses threads). But because Asterisk is only one process, this process is always executed in the same CPU. So we can have a 8 Cores server, with one Core running Asterisk, another Core running operating system stuff/other small daemons and 6 idle cores. Is this correct? Why not? If this is correct, increasing CPU number of Asterisk server box would not increase the performance. I don't see any other process that could use other Cores (like transcoding processes, executing dialplan, etc.) Thank you for your information, -- Carles Pina i Estany GPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona Carles, Asterisk is one process, but as you mentioned multi-threaded as well. Because it is multi-threaded it can run on multiple cores/CPU's at a time. I don't know the internals of Asterisk that well so I can't site specific examples, but I know that there are some scalability bottlenecks people are looking at, specifically with the IAX protocol and how the threads send/receive packets. I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Ryan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] LDAP support
Hello, I've found this information about asterisk and LDAP: http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP which can be out of date. I'm trying this http://www.mezzo.net/asterisk/app_ldap.html however I'm facing the same problems as this unanswered: http://forums.digium.com/viewtopic.php?p=42591sid=05e1d00ab6f9848f4e7b6 39d66cc6d79 Does anybody know how to solve this issue? Moreover I would like to understand if someone is using LDAP (for iax.conf) and with which asterisk plugin (e.g. app_ldap, Asterisk::LDAP Perl module, etc..). Best Regards, Claudio Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Peak number of calls?
Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who wants to know if he's spent too much on the number of ISDN lines he has installed!) Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
On Wednesday 23 January 2008 12:23:24 Gordon Henderson wrote: Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... No, the CDRs would be where that information is stored, if anywhere. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
Gordon Henderson wrote: Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who wants to know if he's spent too much on the number of ISDN lines he has installed!) Cheers, Gordon We use Asterisk-stat from Areski (GPL). It will show peak number of calls by the hour. Select Daily Load, scroll down and choose the hour you want and Fluctuation Graph. Lots of other goodies too. http://areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
Ok good piece software easy on the eyes as they say and I have to say this before I start listing a lot of things that I would love to see, for it to be usable as a good high performance phone. Working with industrial pc switchboards and soft phones of various vendors for some years now, and it all boils down to. How much functionality you can boil into the keyboard. No mouse action should be needed to search a number add an F-key for it. No mouse action should be needed to dial or transfer a number. No mouse action should be needed unless absolutely unavoidable. A_PARTY = caller B_PARTY = operator / called person C_PARTY = number to transferred to STATES: Example to keep it within the numeric key-pad when you receive a call and transfer it. STEP 1 A call is presented. LINE_STATE: Ringing TRANSFER_STATE: inactive TALKING_TO_STATE: inactive STEP 2 Press numeric enter to pick up call. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: inactive TALKING_TO_STATE: A_PARTY STEP 3 Transfer the call Scenario 1: Search out the number in the phonenbook by pressing ex: F10, while talking to the caller, the phone book appears search by name, number or whatever is available and mark the number with arrow keys and dial with NUM-enter. Scenario 2 Press enter a new dial box appears. Type in the number to call. Press enter. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CALLING_C_PARTY TALKING_TO_STATE: DIALBACKTONE STEP 4 The person transferring the call can now make a choice either to do a attended transfer or a blind transfer. Scenario Blind transfer: Simply pressing NUM-enter should do a blind transfer, and the call handling is done and all states are reset, C_PARTY becomes the B_PARTY and so on. The phone is ready for a new call. LINE_STATE: inactive TRANSFER_STATE: inactive TALKING_TO_STATE: inactive Scenario: Attended transfer: The person transferring the call can talk to the C_PARTY LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY Should the operator wish for switching back do the previous call that currently placed on hold it could be done by pressing the NUM+ key placing the C_PARTY on hold and reconnecting the A_PARTY LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: A_PARTY Switch back by NUM+ LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY Connect the call by NUM-enter at any point talking to either the A_PARTY or C_PARTY. The call handling is done and all states are reset, C_PARTY becomes the B_PARTY and so on. The phone is ready for a new call. LINE_STATE: inactive TRANSFER_STATE: inactive TALKING_TO_STATE: inactive Scenario: disconnect the party you are talking to Press NUM- If the states are as follows. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY The C_PARTY would be disconnected and the states would go to. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: inactive TALKING_TO_STATE: A_PARTY And the here we go again with a new transfer or a goodbye and hang up with NUM-. Some side notes: The calling transfer functions are already in the phone alle that needs to be done is associate the functions to the states and numeric keys. The features could be activated by putting the phone in operator mode, if this was the case you could turn of the DTMF and just start typing the new number and hit NUM-enter twice to transfer the call fast. 1 enter to dial number the other to transfer. DTMF could be turned of since the operator rarely calls any ivr, that needs a DTMF response, if so you could leave dtmf open on the QWERTY number keys HEX 30 31 33 34 so on. A tcp port on the phone that allowed for picking up calls and hanging up calls, and perhaps being able to read the number status would make is possible for people write some very nice callcenter agent software for this phone, without having to worry about the functionality of a phone in their agent software. These things might be on the table already if so happy days and then I can't wait to see the product then. Shw that was a little longer than expected. Just my way to keep it simple :), but I hope this could the first really good sip phone with switchboard properties out there. Regards Christian Ejlertsen -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Simon Elliston Ball Sent: 23. januar 2008 13:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer Zoiper is pretty impressive, it's a simple, neat little client. The one problem I have with it is the keyboard. I've had problems trying to use the keyboard to send DTMF on the current call. The left hand popout keypad is also a little small for my users' taste. It
Re: [asterisk-users] Peak number of calls?
On Wed, 23 Jan 2008, Drew Gibson wrote: Gordon Henderson wrote: Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. We use Asterisk-stat from Areski (GPL). It will show peak number of calls by the hour. Select Daily Load, scroll down and choose the hour you want and Fluctuation Graph. Lots of other goodies too. http://areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 Or, as a quick dirty... DATE=$(date +%F-%H-%M-%S) COUNT=$(sudo /usr/sbin/asterisk -r -x sip show channels | wc -l) echo $DATE $COUNT /tmp/channel-counts in a shell script executed every second in cron. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
Gordon Henderson wrote: Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. MRTG is very handy for this. We use the script found at: http://karlsbakk.net/asterisk/ You can plot SIP, IAX, and ZAP Channels over time. Andres http://www.neuroredes.com I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who wants to know if he's spent too much on the number of ISDN lines he has installed!) Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_txfax
Hello, I'm setting up Asterisk to send outgoing faxes over a PRI line. I installed app_txfax and its prerequisites and astfax to submit email messages to Asterisk. This all seems to work fine, but I get some error messages in my logs I don't understand. Whenever I send a fax it goes through fine, but I get these messages in the logs: [Jan 17 11:21:07] WARNING[2413] chan_zap.c: Unable to request echo training on channel 1 [Jan 17 11:21:13] WARNING[2413] pbx.c: Zap/1-1 already has a call record?? [Jan 17 11:21:35] WARNING[2413] /home/sgifford/src/agx-ast-addons/app_txfax.c: Transmission loop error When I send a fax to a line that's busy, I get: [Jan 17 11:40:29] NOTICE[2439] pbx_spool.c: Call failed to go through, reason (0 ) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) while I would expect a simple BUSY. Also, app_txfax will retry the fax a few times before giving up. I'd like to know when it gives up, so I can let the fax sender know that it didn't go through. Is there a way to hook into that? I'm using Asterisk 1.4.17 with spandsp 0.0.4, tx_fax from agx-ast-addons 1.4.3, and astfax 1.0. It's running on Debian Sarge (4.0) with Debian-supplied kernel 2.6.18. I'm using a Digium TE110P card with zaptel driver 1.4.7.1. Thanks for any ideas! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
Hello, On Jan/23/2008, Ryan Burke wrote: I wonder how Asterisk scales when we increment the Core's or CPU's of one computer. I see that Asterisk is only one process (I guess that it uses threads). Asterisk is one process, but as you mentioned multi-threaded as well. Because it is multi-threaded it can run on multiple cores/CPU's at a time. I don't know the internals of Asterisk that well so I can't site specific examples, but I know that there are some scalability bottlenecks people are looking at, specifically with the IAX protocol and how the threads send/receive packets. thanks for information. To give some more details, is we execute: ps auxwm We can see that Asterisk is using quite many threads (33 threads in a mainly new Asterisk installation) I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Yes, I see that it uses threads. I wonder some other data like which is the limit that core/CPU's are correctly used (or usefull used). Thanks again Ryan, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Asterisk is highly multi-threaded and definitely takes advantage of multiple cores. There are a few places where concurrency could be further improved, but its really quite good in 1.4. (IAX in 1.4 does handle traffic using a thread pool so will take advantage of multiple cores). By the way, I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
On Wed, 23 Jan 2008, Stephen Davies wrote: By the way, I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls. With media passing through it? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom 320 Lost Settings
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Has anyone ever seen an Snom320 lose settings? It's been working fine for months and then I got a call this morning saying that it was asking for country, timezone etc. I logged in remotely, and it had lost the server address, username, password, mailbox and ringtone. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHl6vCDQNt8rg0Kp4RAospAJ9DUNge64n7u3RkQWsodHgdOS/higCgwNFy VfZUUNJIgzeC4Hy5vg0f+mY= =tpnK -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 320 Lost Settings
On 23/01/2008, Matt Riddell [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Has anyone ever seen an Snom320 lose settings? It's been working fine for months and then I got a call this morning saying that it was asking for country, timezone etc. I logged in remotely, and it had lost the server address, username, password, mailbox and ringtone. - -- Kind Regards, Matt Riddell Director Yeah I had exactly the same thing. I reported it to Snom and they suggested a procedure for upgrading the firmware, however I have not had chance to do it yet. Mike ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHl6vCDQNt8rg0Kp4RAospAJ9DUNge64n7u3RkQWsodHgdOS/higCgwNFy VfZUUNJIgzeC4Hy5vg0f+mY= =tpnK -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 320 Lost Settings
On 10:04, Thu 24 Jan 08, Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Has anyone ever seen an Snom320 lose settings? It's been working fine for months and then I got a call this morning saying that it was asking for country, timezone etc. I logged in remotely, and it had lost the server address, username, password, mailbox and ringtone. Yup, we have this on some customers from time to time as well. Snom told us the new firmware will fix this but it wont. We sent printed instructions to our customer and they reconfigure the phone once it happens. Very lame. We switched to cisco and aastra phones because this and some other trouble that was discussed earlier this week. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
Hi, On Wed, 2008-01-23 at 16:03 -0500, Alex Balashov wrote: On Wed, 23 Jan 2008, Stephen Davies wrote: By the way, I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls. With media passing through it? if doing conversion from sip 2 iax is pretty difficult to NOT handle media... since iax does not have RTP. regards mat ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 320 Lost Settings
keep in mind that administrative reset is just a few key presses, a bored kid or employee can also cause this.. On Jan 23, 2008 4:04 PM, Matt Riddell [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Has anyone ever seen an Snom320 lose settings? It's been working fine for months and then I got a call this morning saying that it was asking for country, timezone etc. I logged in remotely, and it had lost the server address, username, password, mailbox and ringtone. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHl6vCDQNt8rg0Kp4RAospAJ9DUNge64n7u3RkQWsodHgdOS/higCgwNFy VfZUUNJIgzeC4Hy5vg0f+mY= =tpnK -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] TuxTone Inc. http://www.TuxTone.com */ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
On Jan 23, 2008 2:06 PM, Steve Edwards [EMAIL PROTECTED] wrote: On Wed, 23 Jan 2008, Drew Gibson wrote: Gordon Henderson wrote: Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. We use Asterisk-stat from Areski (GPL). It will show peak number of calls by the hour. Select Daily Load, scroll down and choose the hour you want and Fluctuation Graph. Lots of other goodies too. http://areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 Or, as a quick dirty... DATE=$(date +%F-%H-%M-%S) COUNT=$(sudo /usr/sbin/asterisk -r -x sip show channels | wc -l) echo $DATE $COUNT /tmp/channel-counts in a shell script executed every second in cron. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 I have seen Cacti used to make some *really* nice semi-realtime and historic graphs. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
Sorry to be a little OT.. But may I ask what some more of the specs are for that machine? Just trying to get an idea of what different hardware can achieve. Thanks, Daniel From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Davies Sent: Thursday, 24 January 2008 7:57 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk scalability I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Asterisk is highly multi-threaded and definitely takes advantage of multiple cores. There are a few places where concurrency could be further improved, but its really quite good in 1.4. (IAX in 1.4 does handle traffic using a thread pool so will take advantage of multiple cores). By the way, I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Replacement for Allison
Hi, Does anyone know what I need to do to get these: http://www.enicomms.com/cutglassivr/ Sounds files to work? I've tried loading them, but they are completely silent (format mis-match maybe?). Specifically, when I try to enter voicemail, nothing plays... though it clearly tries. I'm looking for replacement sound files for the default Allison, as I feel she is kind of breathy. I have heard other sound files on other asterisk sounds, done by her, and they sound fine... are there two recorded versions of the prompts floating around? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
On Wed, 23 Jan 2008, Andres wrote: Gordon Henderson wrote: Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. MRTG is very handy for this. We use the script found at: http://karlsbakk.net/asterisk/ You can plot SIP, IAX, and ZAP Channels over time. Ah yes. Quite Intersting. I use MRTG in a lot of applications, so this is worthy of a look. It only samples every 5 minutes though, so has the potential to miss things, although this (and the crude shell-script suggested by Steve Edwards has presented me with an idea to use the manager interface to sample it a bit more often and keep a count. Thanks! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
Thank you very much for the feedback, i definately like the suggestions and i will do my best to get this on the roadmap. (which should be pretty easy as i actually kind of make the roadmap :p), so expect in done in one of the following releases. The things to turn it into a callcenter application are already there, not with a TCP port, but you could use it with command line options (even if the phone is already running) or through a com object. Documentation can be found here: http://www.zoiper.com/downloads/Zoiper_API_Documentation.pdf Examples can be found here : http://www.zoiper.com/biz3.php I have an example for jscript somewhere tool, contact me offlist if you want it. Let me know offlist if you need any biz licenses to try it out, i;d be happy to provide you with them. Zoa. Christian Ejlertsen wrote: Ok good piece software easy on the eyes as they say and I have to say this before I start listing a lot of things that I would love to see, for it to be usable as a good high performance phone. Working with industrial pc switchboards and soft phones of various vendors for some years now, and it all boils down to. How much functionality you can boil into the keyboard. No mouse action should be needed to search a number add an F-key for it. No mouse action should be needed to dial or transfer a number. No mouse action should be needed unless absolutely unavoidable. A_PARTY = caller B_PARTY = operator / called person C_PARTY = number to transferred to STATES: Example to keep it within the numeric key-pad when you receive a call and transfer it. STEP 1 A call is presented. LINE_STATE: Ringing TRANSFER_STATE: inactive TALKING_TO_STATE: inactive STEP 2 Press numeric enter to pick up call. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: inactive TALKING_TO_STATE: A_PARTY STEP 3 Transfer the call Scenario 1: Search out the number in the phonenbook by pressing ex: F10, while talking to the caller, the phone book appears search by name, number or whatever is available and mark the number with arrow keys and dial with NUM-enter. Scenario 2 Press enter a new dial box appears. Type in the number to call. Press enter. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CALLING_C_PARTY TALKING_TO_STATE: DIALBACKTONE STEP 4 The person transferring the call can now make a choice either to do a attended transfer or a blind transfer. Scenario Blind transfer: Simply pressing NUM-enter should do a blind transfer, and the call handling is done and all states are reset, C_PARTY becomes the B_PARTY and so on. The phone is ready for a new call. LINE_STATE: inactive TRANSFER_STATE: inactive TALKING_TO_STATE: inactive Scenario: Attended transfer: The person transferring the call can talk to the C_PARTY LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY Should the operator wish for switching back do the previous call that currently placed on hold it could be done by pressing the NUM+ key placing the C_PARTY on hold and reconnecting the A_PARTY LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: A_PARTY Switch back by NUM+ LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY Connect the call by NUM-enter at any point talking to either the A_PARTY or C_PARTY. The call handling is done and all states are reset, C_PARTY becomes the B_PARTY and so on. The phone is ready for a new call. LINE_STATE: inactive TRANSFER_STATE: inactive TALKING_TO_STATE: inactive Scenario: disconnect the party you are talking to Press NUM- If the states are as follows. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY The C_PARTY would be disconnected and the states would go to. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: inactive TALKING_TO_STATE: A_PARTY And the here we go again with a new transfer or a goodbye and hang up with NUM-. Some side notes: The calling transfer functions are already in the phone alle that needs to be done is associate the functions to the states and numeric keys. The features could be activated by putting the phone in operator mode, if this was the case you could turn of the DTMF and just start typing the new number and hit NUM-enter twice to transfer the call fast. 1 enter to dial number the other to transfer. DTMF could be turned of since the operator rarely calls any ivr, that needs a DTMF response, if so you could leave dtmf open on the QWERTY number keys HEX 30 31 33 34 so on. A tcp port on the phone that allowed for picking up calls and hanging up calls, and perhaps being able to read the number status would make is possible for people write some very nice callcenter agent software for this phone, without having to worry about
[asterisk-users] nokia e51 (Christian Lox)
Hi Christian, I have been using the Nokia E51 with asterisk for a month now without any problems. It took me a while to configure it. I downloaded from Nokia a file (dont remember the name now, I am not on my pc at them moment) that added more features such as g729 etc. it is working great. My asterisk is on a public ip address, maybe that helps. Take care, -- Sergio Fabian Veltri Director Business IT Of: +54-11-5217-1297 Ext. 2201 Cell: +54-911-5977-0977 http://www.businessit.biz IT Service Management and Control Best Practices -- Message: 5 Date: Sun, 20 Jan 2008 00:10:58 +0100 From: Christian Lox [EMAIL PROTECTED] Subject: [asterisk-users] nokia e51 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-15; format=flowed Hi all. Anyone to share some experience with Nokia E51 and asterisk? We are trying to connect the E51 to our asterisk but to no avail. Googling said that it should work, but we are seeing real strange things here: - tcpdump reveals the nokia is talking to other ports than 5060 - registration is not possible at all, right now there is no network traffic to the asterisk box at all. A softphone on the same wlan segment registers without any problem. The how-tos on the web suggest different settings concerning the proxy/registration setupBut none of them works for us. But we are not nokia guys at all So, any help greatly appreciated! The setup: Cisco AP with EAP-TLS. Connected to an switch on which several vlans are connected to a cisco router. The internal network (192.168.23.0/24) talks to the DMZ, on which the radius (for EAP-TLS) and also the asterisk box is hosted. IP Addresses are assigned via DHCP from the AP. The Laptop from which i am writing has x-lite installed and that works just fine with the same credentials we are trying to setup the nokia: 2001 abc sipgate No RFC3581 We have been playing with nat=yes|no, but we cant get it to work. Thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
There was a cool paper written a a few months ago where they tested some older dell servers - full details of specs and tests were available. PaulH On Thu, 2008-01-24 at 08:54 +1100, Daniel Cole wrote: Sorry to be a little OT.. But may I ask what some more of the specs are for that machine? Just trying to get an idea of what different hardware can achieve. Thanks, Daniel __ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Davies Sent: Thursday, 24 January 2008 7:57 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk scalability I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Asterisk is highly multi-threaded and definitely takes advantage of multiple cores. There are a few places where concurrency could be further improved, but its really quite good in 1.4. (IAX in 1.4 does handle traffic using a thread pool so will take advantage of multiple cores). By the way, I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
On Wed, 2008-01-23 at 18:23 +, Gordon Henderson wrote: Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who wants to know if he's spent too much on the number of ISDN lines he has installed!) Munin has a nice Asterisk plugin that works reasonably well. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem bridging on Asterisk (no VoIP involved)
On Wed, 2008-01-23 at 09:39 +0100, Alberto Pastore wrote: Hi everybody. I know maybe this question has been posted some time ago, but I need your updated opinion on the subject. I'm replacing our old pbx with asterisk. I have two TE207 dual pri (e1) cards on a clustered system (one on each node). I absolutely need to connect 4/5 analog extensions with modems, they're being used for remote assistance on very old systems which cannot be upgraded to native IP links. Is there a good hardware that can bridge the e1 lines on the digium te207 card to my modems? A PCI card? An external box? I don't want to relay modem connections over ip, I just need to bridge them internally on the asterisk server: E1 == TE207 == Asterisk == (some hardware with FXS) == modems TIA for your replies. Xorcom make things that work for this56k might not be attainable, but lower speeds definitely are. http://www.xorcom.com/ PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
Link? Thanks, Steve Totaro On Jan 23, 2008 6:08 PM, Paul Hales [EMAIL PROTECTED] wrote: There was a cool paper written a a few months ago where they tested some older dell servers - full details of specs and tests were available. PaulH On Thu, 2008-01-24 at 08:54 +1100, Daniel Cole wrote: Sorry to be a little OT.. But may I ask what some more of the specs are for that machine? Just trying to get an idea of what different hardware can achieve. Thanks, Daniel __ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Davies Sent: Thursday, 24 January 2008 7:57 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk scalability I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Asterisk is highly multi-threaded and definitely takes advantage of multiple cores. There are a few places where concurrency could be further improved, but its really quite good in 1.4. (IAX in 1.4 does handle traffic using a thread pool so will take advantage of multiple cores). By the way, I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Parking with multiple lots
Hi List, I need to have one PBX but have multiple call parking for many different context. Basically for hosted VoIP, anyway this can be achineved? We really want to use the Snom's or something like that with a light on the phone so we can what caller is in each parking space/line. I have not seen anyway to do this, any ideals anyone? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Parking with multiple lots
Look at app_valetparking, available here: http://www.freeswitch.org/asterisk_stuff/ I do not know about phone notification (I just use ringback/overhead paging), but it handles multiple contexts just fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthy Sent: Wednesday, January 23, 2008 15:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call Parking with multiple lots Hi List, I need to have one PBX but have multiple call parking for many different context. Basically for hosted VoIP, anyway this can be achineved? We really want to use the Snom's or something like that with a light on the phone so we can what caller is in each parking space/line. I have not seen anyway to do this, any ideals anyone? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
Steve Edwards wrote: Or, as a quick dirty... DATE=$(date +%F-%H-%M-%S) COUNT=$(sudo /usr/sbin/asterisk -r -x sip show channels | wc -l) echo $DATE $COUNT /tmp/channel-counts in a shell script executed every second in cron. every *second* from cron? how the heck would I you do that? sub-minute accuracy from cron is something I don't know how to do. Maybe it's a different version of cron...? The only way I would achieve that would be to run something every minute that self-perpetuated for the rest of that minute... for x in `seq 1 58`; do ( DATE=$(date +%F-%H-%M-%S) COUNT=$(sudo /usr/sbin/asterisk -r -x sip show channels | wc -l) echo $DATE $COUNT /tmp/channel-counts ) sleep 1s done which is honestly very messy. I promise I'm not being sarcastic. I actually *am* curious if there are versions of cron that will go sub-minute. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
http://www.transnexus.com/White% 20Papers/asterisk_V1-4-11_performance.htm It was the bottom news item on voip-info.org - I was worried I would have to really search for it! later, PaulH On Wed, 2008-01-23 at 18:30 -0500, Steve Totaro wrote: Link? Thanks, Steve Totaro On Jan 23, 2008 6:08 PM, Paul Hales [EMAIL PROTECTED] wrote: There was a cool paper written a a few months ago where they tested some older dell servers - full details of specs and tests were available. PaulH On Thu, 2008-01-24 at 08:54 +1100, Daniel Cole wrote: Sorry to be a little OT.. But may I ask what some more of the specs are for that machine? Just trying to get an idea of what different hardware can achieve. Thanks, Daniel __ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Davies Sent: Thursday, 24 January 2008 7:57 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk scalability I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Asterisk is highly multi-threaded and definitely takes advantage of multiple cores. There are a few places where concurrency could be further improved, but its really quite good in 1.4. (IAX in 1.4 does handle traffic using a thread pool so will take advantage of multiple cores). By the way, I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
On Wed, 23 Jan 2008, Mojo with Horan Company, LLC wrote: Steve Edwards wrote: Or, as a quick dirty... DATE=$(date +%F-%H-%M-%S) COUNT=$(sudo /usr/sbin/asterisk -r -x sip show channels | wc -l) echo $DATE $COUNT /tmp/channel-counts in a shell script executed every second in cron. every *second* from cron? how the heck would I you do that? sub-minute accuracy from cron is something I don't know how to do. Sheese -- that's what I get by trying to type without putting down the crack pipe :) You're right -- the * in the first column of your crontab means minutes, not seconds. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Parking with multiple lots
How many contexts have you had this running on? And for the ring back, you cant have it park and then on the same call return the info, has to hangup then ring back? Thanks! On Jan 23, 2008 4:48 PM, Darryl Dunkin [EMAIL PROTECTED] wrote: Look at app_valetparking, available here: http://www.freeswitch.org/asterisk_stuff/ I do not know about phone notification (I just use ringback/overhead paging), but it handles multiple contexts just fine. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Ron McCarthy *Sent:* Wednesday, January 23, 2008 15:39 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Call Parking with multiple lots Hi List, I need to have one PBX but have multiple call parking for many different context. Basically for hosted VoIP, anyway this can be achineved? We really want to use the Snom's or something like that with a light on the phone so we can what caller is in each parking space/line. I have not seen anyway to do this, any ideals anyone? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Parking with multiple lots
I've had two live, it's a pretty archaic feature that emulates older PBXs so it isn't a popular feature at all. Just check the source on your options: -= Info about application 'ValetParkCall' =- [Synopsis] Valet Park Call [Description] ValetParkCall(exten|lotname|timeout[|return_ext][|return_pri][ |return_context]) Park Call at exten in lotname until someone calls ValetUnparkCall on the same exten + lotname set exten to 'auto' to auto-choose the slot. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthy Sent: Wednesday, January 23, 2008 16:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Parking with multiple lots How many contexts have you had this running on? And for the ring back, you cant have it park and then on the same call return the info, has to hangup then ring back? Thanks! On Jan 23, 2008 4:48 PM, Darryl Dunkin [EMAIL PROTECTED] wrote: Look at app_valetparking, available here: http://www.freeswitch.org/asterisk_stuff/ I do not know about phone notification (I just use ringback/overhead paging), but it handles multiple contexts just fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthy Sent: Wednesday, January 23, 2008 15:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call Parking with multiple lots Hi List, I need to have one PBX but have multiple call parking for many different context. Basically for hosted VoIP, anyway this can be achineved? We really want to use the Snom's or something like that with a light on the phone so we can what caller is in each parking space/line. I have not seen anyway to do this, any ideals anyone? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
Thank you Paul! Its impressive! On Jan 23, 2008 4:55 PM, Paul Hales [EMAIL PROTECTED] wrote: http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm It was the bottom news item on voip-info.org - I was worried I would have to really search for it! later, PaulH On Wed, 2008-01-23 at 18:30 -0500, Steve Totaro wrote: Link? Thanks,Steve Totaro On Jan 23, 2008 6:08 PM, Paul Hales [EMAIL PROTECTED] wrote: There was a cool paper written a a few months ago where they tested some older dell servers - full details of specs and tests were available. PaulH On Thu, 2008-01-24 at 08:54 +1100, Daniel Cole wrote: Sorry to be a little OT.. But may I ask what some more of the specs are for that machine? Just trying to get an idea of what different hardware can achieve. Thanks,Daniel __ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Davies Sent: Thursday, 24 January 2008 7:57 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk scalability I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Asterisk is highly multi-threaded and definitely takes advantage of multiple cores.There are a few places where concurrency could be further improved, but its really quite good in 1.4. (IAX in 1.4 does handle traffic using a thread pool so will take advantage of multiple cores).By the way, I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls. Steve ___-- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
Steve Edwards wrote: in a shell script executed every second in cron. every *second* from cron? how the heck would I you do that? sub-minute accuracy from cron is something I don't know how to do. Sheese -- that's what I get by trying to type without putting down the crack pipe :) You're right -- the * in the first column of your crontab means minutes, not seconds. Ok, I'm NOT on the crack pipe then ;) I was wondering. Sticking to the slimy hack i described! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Parking with multiple lots
I agree it is old, some people won't adopt. We run into this with clients who are to use to legacy key systems. I have found no other real way around this when you need this feature, some way for another person in a office to pick up a call. Its a hassle, wish some people would change! On 1/23/08, Darryl Dunkin [EMAIL PROTECTED] wrote: I've had two live, it's a pretty archaic feature that emulates older PBXs so it isn't a popular feature at all. Just check the source on your options: -= Info about application 'ValetParkCall' =- [Synopsis] Valet Park Call [Description] ValetParkCall(exten|lotname|timeout[|return_ext][|return_pri][ |return_context]) Park Call at exten in lotname until someone calls ValetUnparkCall on the same exten + lotname set exten to 'auto' to auto-choose the slot. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthy Sent: Wednesday, January 23, 2008 16:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Parking with multiple lots How many contexts have you had this running on? And for the ring back, you cant have it park and then on the same call return the info, has to hangup then ring back? Thanks! On Jan 23, 2008 4:48 PM, Darryl Dunkin [EMAIL PROTECTED] wrote: Look at app_valetparking, available here: http://www.freeswitch.org/asterisk_stuff/ I do not know about phone notification (I just use ringback/overhead paging), but it handles multiple contexts just fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthy Sent: Wednesday, January 23, 2008 15:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call Parking with multiple lots Hi List, I need to have one PBX but have multiple call parking for many different context. Basically for hosted VoIP, anyway this can be achineved? We really want to use the Snom's or something like that with a light on the phone so we can what caller is in each parking space/line. I have not seen anyway to do this, any ideals anyone? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replacement for Allison
for x in *.g711u; do mv $x ${x%.g711u}.ulaw; done
On Jan 23, 2008 5:00 PM, Matt [EMAIL PROTECTED] wrote:
Hi,
Does anyone know what I need to do to get these:
http://www.enicomms.com/cutglassivr/
Sounds files to work? I've tried loading them, but they are completely
silent (format mis-match maybe?). Specifically, when I try to enter
voicemail, nothing plays... though it clearly tries.
I'm looking for replacement sound files for the default Allison, as I feel
she is kind of breathy. I have heard other sound files on other asterisk
sounds, done by her, and they sound fine... are there two recorded
versions of the prompts floating around?
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[asterisk-users] Your favorite Asterisk application.
Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? Thanks, -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Your favorite Asterisk application.
I love writing dialplan, using vi. Does that make me weird? PaulH On Wed, 2008-01-23 at 23:57 -0500, Ken D'Ambrosio wrote: Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? Thanks, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Your favorite Asterisk application.
The fact that it is so amazingly configurable should be enough :) -Kev Ken D'Ambrosio wrote: Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? Thanks, -Ken -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
Tilghman Lesher wrote: On Wednesday 23 January 2008 12:23:24 Gordon Henderson wrote: Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... No, the CDRs would be where that information is stored, if anywhere. This is actually sort of easy. You simply have every call pass through a context in which you assign the call to a group, then either do a NoOp echoing the group count or a user event doing the same, then either programatically or grep search your logs for the output or have a script monitoring the AMI watch for the user event and write the number in a data base. Of these two I personally do the second option because then I can just do a max() function on that database field to get the maximum calls for any time range I specify. Oh and just a note, never just say no because you don't know, in this instance you would say, I think your best bet is the CDR's. Just a tip. Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Your favorite Asterisk application.
Paul Hales wrote: I love writing dialplan, using vi. Does that make me weird? PaulH On Wed, 2008-01-23 at 23:57 -0500, Ken D'Ambrosio wrote: Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? Thanks, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I simply love vi, to the point which if an IDE doesn't have vi key bindings I loath using it. Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
On Wednesday 23 January 2008 23:23:23 Anthony Francis wrote: Tilghman Lesher wrote: On Wednesday 23 January 2008 12:23:24 Gordon Henderson wrote: Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... No, the CDRs would be where that information is stored, if anywhere. This is actually sort of easy. You simply have every call pass through a context in which you assign the call to a group, then either do a NoOp echoing the group count or a user event doing the same, then either programatically or grep search your logs for the output or have a script monitoring the AMI watch for the user event and write the number in a data base. Of these two I personally do the second option because then I can just do a max() function on that database field to get the maximum calls for any time range I specify. Oh and just a note, never just say no because you don't know, in this instance you would say, I think your best bet is the CDR's. Just a tip. The key phrase in the original post was has had, indicating past behavior, not future behavior. Yes, you can do all sorts of things in the dialplan to get that information into a logfile, but you cannot retroactively do those things. The only place that information can be had are the CDRs, so I will stick with my original assessment. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM800P FXO problem incomming call
On Jan 22, 2008 1:50 PM, satish patel [EMAIL PROTECTED] wrote: Dear all I have asterisk 1.4.11 on Cent 4.3 i have faceing some problem i have TDM800P 8 port FXO card when i terminate PSTN line on this port can make outgoing call it is working fine but incomming call not handling ...when i call from outside to this line it is rinning but no one call land on my asterisk no debug in asterisk some time it land but most of time not . check the dialplan to match or contact provider for the problem. simple solution..take the line connect to phone ( if not e1), check incoming call coming or not ? ram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Your favorite Asterisk application.
Yes, but I already knew that. :) Paul Hales wrote: I love writing dialplan, using vi. Does that make me weird? PaulH On Wed, 2008-01-23 at 23:57 -0500, Ken D'Ambrosio wrote: Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? Thanks, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk optimalization
http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm check this link may help you ram On Jan 23, 2008 10:23 PM, marek cervenka [EMAIL PROTECTED] wrote: hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is major bottleneck? qualify imho not. i'm tried set qualify=no, does not help SIP REGISTER packets? this problem persist if no calls are active after restart cpu usage slowly increase. after a ~hour is about 100% which optimalizations do you recommend for ~1500 peers scenario? (behind nat, reregistrations) --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Your favorite Asterisk application.
So I'm not the only one! Kev Anthony Francis wrote: Paul Hales wrote: I love writing dialplan, using vi. Does that make me weird? PaulH On Wed, 2008-01-23 at 23:57 -0500, Ken D'Ambrosio wrote: Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? Thanks, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I simply love vi, to the point which if an IDE doesn't have vi key bindings I loath using it. Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Your favorite Asterisk application.
Asterisk really comes into it's own with cute scripts that can do almost anything with ridiculous ease. One of the things I've done with a number of Asterisk machines is to put in a script that downloads the latest weather forecast and reads it back to you using a TTS engine. Ken D'Ambrosio wrote: Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Your favorite Asterisk application.
With comments like that people are going to think that we aren't related. PaulH On Thu, 2008-01-24 at 16:46 +1100, Rob Hillis wrote: Yes, but I already knew that. :) Paul Hales wrote: I love writing dialplan, using vi. Does that make me weird? PaulH On Wed, 2008-01-23 at 23:57 -0500, Ken D'Ambrosio wrote: Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? Thanks, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk optimalization
ram wrote: http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm check this link may help you ram On Jan 23, 2008 10:23 PM, marek cervenka [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is major bottleneck? qualify imho not. i'm tried set qualify=no, does not help SIP REGISTER packets? this problem persist if no calls are active after restart cpu usage slowly increase. after a ~hour is about 100% which optimalizations do you recommend for ~1500 peers scenario? (behind nat, reregistrations) --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm That result is suprising! but i have DELL 2950 with 2 X 3.0GHz CPU on 6GB ram, equiped with 8e1 link (2 sangoma A104D) running FC5. I installed chan_ss7-1.0 with asterisk-1.2.25 doing transcoding, and each time calls get to 120+ the cpu is fully utilized. the calls come from sip to the ss7 link. can someone advice me on what I can do to improve the performance. goksie NB. I felt we re talking on the same topic thats why i added my own experience. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Your favorite Asterisk application.
You know full well I'm not related to you - I just work with you. :) Paul Hales wrote: With comments like that people are going to think that we aren't related. PaulH On Thu, 2008-01-24 at 16:46 +1100, Rob Hillis wrote: Yes, but I already knew that. :) Paul Hales wrote: I love writing dialplan, using vi. Does that make me weird? PaulH On Wed, 2008-01-23 at 23:57 -0500, Ken D'Ambrosio wrote: Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? Thanks, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] two zaptel card
hi, all I want to use two zaptel card(TE210p) in pc for asterisk. Is there any special requirement for this configuratin. any suggestion. thanks , Bhrugu Mehta ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two zaptel card
Hi , No need of any special requierments. On Jan 24, 2008 12:52 PM, Bhrugu Mehta [EMAIL PROTECTED] wrote: hi, all I want to use two zaptel card(TE210p) in pc for asterisk. Is there any special requirement for this configuratin. any suggestion. thanks , Bhrugu Mehta ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk optimalization
Hi, Dell is not a recomeded server for linux. Its only compatible with windows. On Jan 24, 2008 12:02 PM, Goke Aruna [EMAIL PROTECTED] wrote: ram wrote: http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm check this link may help you ram On Jan 23, 2008 10:23 PM, marek cervenka [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is major bottleneck? qualify imho not. i'm tried set qualify=no, does not help SIP REGISTER packets? this problem persist if no calls are active after restart cpu usage slowly increase. after a ~hour is about 100% which optimalizations do you recommend for ~1500 peers scenario? (behind nat, reregistrations) --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm That result is suprising! but i have DELL 2950 with 2 X 3.0GHz CPU on 6GB ram, equiped with 8e1 link (2 sangoma A104D) running FC5. I installed chan_ss7-1.0 with asterisk-1.2.25 doing transcoding, and each time calls get to 120+ the cpu is fully utilized. the calls come from sip to the ss7 link. can someone advice me on what I can do to improve the performance. goksie NB. I felt we re talking on the same topic thats why i added my own experience. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
