Re: [Asterisk-Users] Polycom Delay
Hey everyone, Hopefully someone can point me in the right direction for this. Currently we have two offices, all using Polycom 601 Revsion E I think. All have the same configurations and firmware versions. The differences: Office A: public IP address. Office B: NAT (router has a static IP) Office A: Same state as the asterisk server (Michigan) Office B: Wisconsin Office A: T1 network to the colo where the asterisk server is located Office B: Wireless connection (2 tower hops I think) (our wireless connection, we are a small ISP) to our backbone to the colo Okay, so calls going to and from office A have no problems at all. Office B is having a bit of a delay (about 5 seconds before the CLI shows the call is even started). The odd part is, it only happens when they are making an outbound call. Incoming calls go directly to them without any problems. Both offices for external calls use our PRI we have installed and all interal are SIP. I think also internal calls are having the same problem, but that I haven't had a 100% sure answer if it is or isn't, but I know for sure the PRI calls are. My question is, does it sound like the phone is causing the problem, or the network being NAT, wireless connection, or both having more to do with the problem. While I know it isn't an answer you can say, hey this is the solution, I would like any input or experience that anyone has had with a problem like this. Thanks Kevin I have two thoughts. First, what is the latency between Office B and your * box at the colo? What is the latency between office A and your * box? Compare these two and see if there is any corrilation to the delay when making a call from office B. If so, then its unfortunetly your network configuration (using wireless network). I know this isn't very useful, but hopefully its at least a start in some direction. Another thing which is probably vry unlikely (but who knows)...maybe the phones at office B are not setup with a digit map and they are having to wait for the DTMF timeout to be reached before sending the call? Silly, I know. - Gabe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Development news :: New AEL and configuration system
Friends in the Asterisk community, Yesterday the Asterisk development branch, also known as svn trunk, changed quite a lot. We added two major features: A new version of AEL and a new configuration system. Hang on, and I'll explain! * AEL - The Asterisk Extension Language Last summer, Mark Spencer created a new language for creating your Asterisk dial plan. Before that, many developers tried making the current dial plan language into a script language by adding if/then/else and do/while constructs - and it all seemed very strange and, well, not really like a script language. So Mark decided to take another route and implemented a new language, that was interpreted into the old. You could suddenly create a dial plan in a language that looked more like C, and let the AEL parser create a dial plan based on the old language. This first version was experimental and had a lot of problems. Writing a language parser is not an easy task. Remember that what you write in the AEL file and what you see when you do show dialplan in the CLI is very different. AEL is still interpreted into the old dial plan language. The new AEL is implemented using Bison, which leads to a much more robust parser. Steve Murphy has put a lot of work into implementing AEL2 and it looks very good. So good, so Kevin removed the experimental flag on AEL, making it a standard feature in Asterisk. * AUTOCONF and MENUSELECT - Installation now is easier! Since I joined the Asterisk community, I have seen regular requests for a ./configure script for Asterisk. The Asterisk Makefile replaced some of the functionality of the ./configure script, trying to find out what functionality was available on the host system. Yesterday, we finally got an auto-configuration system. The Makefile now creates a configure script, runs it to check what you have - MySQL, OSP, PostgreSQL, CURL etc - and make sure the optimal Asterisk is created on your system. Additionally, you can run make menuselect to be able to select what modules you want. No app_dial.so? Just disable it! Menuselect also marks clearly modules that can't be installed on your system due to lacking third party libraries. And to top it off, we now have ASCII art embedded into Asterisk! * Making life easier for the Asterisk administrator -- While these additions does not really change the functionality of your favourite PBX, they make installation and configuration of your Asterisk system easier. It's a big step forward and an important part of Asterisk 1.4. Now, I have to learn the inner workings of this and adopt my branches to it... Always good to have something to do ;-) Greetings from the Asterisk Developer Community! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk European tour - REGISTER NOW - http://www.meetasterisk.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Two asterisk process in one hardware.
On Tuesday 25 April 2006 00:57, Juan Salas wrote: Hello. Has anybody knows how run two asterisk process in one hardware? (each one with its own configuration?) It is possible. 1) Use different UDP ports for SIP/IAX/RTP 2) Use different log files and astdb files But most users do not need this. If you need separate phone systems for multiple customers, use separate contexts instead. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk2Billing
Scheda wrote: I'm sure this has been asked a million times. Therefore, I must ask again. Generally speaking, what do you guys think of it. It looks pretty good, but for my uses, I'm not sure that a calling card method is the *best* way to go. But, either way, what is the general concensus? Rock stable, and IMHO the best solution atm. been using for half a year, never had a problem. Kudos to Areski. Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Two asterisk process in one hardware.
On Tue, 25 Apr 2006, Dmitry Ivanov wrote: On Tuesday 25 April 2006 00:57, Juan Salas wrote: Hello. Has anybody knows how run two asterisk process in one hardware? (each one with its own configuration?) It is possible. 1) Use different UDP ports for SIP/IAX/RTP This is not necessary if you have two network interfaces to bind the two asterisks on. 2) Use different log files and astdb files One asterisk could also run in a chroot environment (or even both). Armin But most users do not need this. If you need separate phone systems for multiple customers, use separate contexts instead. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CHANUNAVAIL, busy and congestion
Greetings to all, I ma having a problem with channel variables on a couple of our Asterisk boxes. Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN. On the External GW, we also have an IAX trunk to a VOIP provider if for some reason the E1 is down. If the DIALSTATUS is CHANUNAVAIL, which should be returned from the PSTN GW if the E1 is unavailable, the call goes out over our VOIP provider. Below is an example of a call that goes tires the PSTN GW, gets a DIALSTATUS of CHANUNAVAIL, and then calls our VOIP provider. The problem is that this call was actually busy, and the E1 was not unavailable. -- Called RemoteServ/0089538881220* -- Call accepted by 172.16.10.2 (format alaw) -- Format for call is alaw -- Hungup 'IAX2/RemoteServ-10' == Everyone is busy/congested at this time (1:0/0/1) -- Executing NoOp(IAX2/sanset-5, CHANUNAVAIL) in new stack -- Executing Goto(IAX2/sanset-5, s-CHANUNAVAIL|1) in new stack -- Goto (sansetuplink,s-CHANUNAVAIL,1) -- Executing Dial(IAX2/sanset-5, IAX2/munich:[EMAIL PROTECTED]/0089538881220*) in new stack -- Called munich:[EMAIL PROTECTED]/0089538881220* -- Call accepted by 202.148.48.242 (format gsm) -- Format for call is gsm -- IAX2/london-15 is making progress passing it to IAX2/sanset-5 -- IAX2/london-15 is ringing -- IAX2/london-15 stopped sounds -- IAX2/london-15 answered IAX2/sanset-5 If anyone has any ideas why this is not working as expected please let me know. Or anything I can do to try and solve this. I've also experienced the same thing calling numbers that do not exist. Thanks, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] billing realtime
Hi all I think this could be en old question. I would like to do a realtime billing prepaid system, mainly using asterisk. I have found few things; I can not get CDR function into agi because asterisk set them once the call is absolutely finish (at least main values for the main porpouse, billsec,duration, etc..) There is a patch that allow you to use CDR function in hangup extension, but it seems to have some troubles, haven't it?? Finally, another solution I have found reading somewhere could be use triggers in the database when there is and CDR insert. A cron job could do something similar. Now, the question, can I access somehow in a deadagi, or whatever the CDR function in order to update the credit when the call has just finished. Thank you very much ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialing Ring Groups from the Digital Receptionist-
Hi, I only check the AAH AMP. The inbound routing from-pstn didn't include the context ext-group. So the ring group setting doesn't work when you call from PSTN. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, April 25, 2006 11:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dialing Ring Groups from the Digital Receptionist- Hi! I've got a number of extensions (about 50) on a working Asterisk setup. For each user, I have two extensions configured (for example 11021 for a Cisco 79XX phone and 11022 for X-Lite), and a ring group that ties the two extensions together (for example, 1102). Reason being that if the user is away from his/her desk or working offsite, they can answer the soft phone on the PC. From an inside SIP extension (say 11071) I can dial 1102 and have it ring both 11021 and 11022, and this setup works well. But when I call the external number and get the digital receptionist, I cannot dial 1102 and have it ring both extensions - I have to either specify 11021 or 11022. So my questions: 1. Clearly it is possible to setup an option in the digital receptionist and have it dial 1102 (press 3 for Bob - dial 1102), but this doesn't scale well for 50 users. So is there a way to dial 1102 from the digital receptionist and have it ring both 11021 and 11022? 2. Is there another way to accomplish what I'm trying to do, ie. have two extensions per user, then dial them both simultaneously, and leave it up to the user to decide which one to answer - and do this from a phone NOT connected to the VoIP system? I appreciate your responses. Thanks- --Maxx ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] wellgate FXO unit
Yes, just set the hotline number to an extension number. And disable the welltech IVR function. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Artifex Maximus Sent: Tuesday, April 25, 2006 3:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] wellgate FXO unit May hotline function will help. I never been use with Asterisk just with Welltech FXS device so it's just a hint. artifex On 4/21/06, Jerry Geis [EMAIL PROTECTED] wrote: Anyone know how to set the wellgate unit so incoming calls pass on directly to asterisk? Right now incoming calls ring twice and I hear a recording saying enter the extension. If I go enter the extension it goes on to asterisk just fine. I just want the incoming call to go directly onto asterisk. Anyone found that out? Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)
25 apr 2006 kl. 00.24 skrev Thomas Winter: Am Monday 24 April 2006 18:39 schrieb Doug Lytle: Thomas Winter wrote: Hi, I dont want to have in the SIP HEADER the CALLERID(name) (the Display Name) for the initial INVITE to an SIP proxy. If I use SET(CALLERID(name)=) the display-name is asterisk. Just a guess, try: SET(CALLERID(name)= ) Hi, Asterisk will use this space. - FROM: sip:CALLERID(number)@domain.tld We do insert asterisk when we have no caller ID name. In what situations don't you want a caller ID name at all? I am a bit curious here, in order to understand. Regards /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 testing
HiI sent this earlier, but it was late and I haven't saw any reply. Maybe now I will have more luckDoes anyone know the correct settings of zapata.conf and zaptel.conf that are needed to connect two asterisk boxes over E1. I am trying to (just for testing purposes) connect two * ( A and B ) boxes over E1 link and IAX as well. Both are Soekris 4801 and have Sangma A101U cards. The situation looks like this: I have a Sip phone connected to Asterisk A. The call goes from Asterisk A to Asterisk B over E1 link then it goes back to Asterisk A over IAX then once again to Asterisk B over E1 and back to Asterisk A over IAX and so on . I want to use all 30 channels of E1 but something is just not right. Asterisk hangs up all channels after making third loop. Is it possible to make such a loop in asterisk or maybe it is internally protected from doing it? Or maybe I configure it in a wrong way? My zapata and zaptel conf looks like this: zaptel.conf:loadzone=nldefaultzone=nlspan=1,0,0,ccs,hdb3,crc4 # span=1,1,0,ccs,hdb3,crc4 in the other asterisk boxbchan=1-15, 17-31dchan=16zapata.conf:[channels]context=soekris switchtype=euroisdnpridialplan=unknownprilocaldialplan=unknownsignalling=pri_net ;;pri_cpe in the other oneusecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yescallwaitingcallerid=yes threewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0group=1callgroup=1pickupgroup=1immediate=nochannel = 1-15 channel = 17-31Logs are available in txt attachmentDoes anyone have any cluesThanks in advance Cheers Andrew Console logs from Asterisk A: Executing Dial(SIP/test0-5821, Zap/6/327557670||Tt) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 6/327557670 -- Zap/6-1 is proceeding passing it to SIP/test0-5821 -- Accepting UNAUTHENTICATED call from 195.66.73.122: requested format = alaw, requested prefs = (alaw|gsm), actual format = alaw, host prefs = (alaw|gsm), priority = mine -- Executing Set(IAX2/soekris2-1, CALLERID(number)=0327557574) in new stack -- Executing Dial(IAX2/soekris2-1, Zap/1/327557671||tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1/327557671 -- Zap/1-1 is proceeding passing it to IAX2/soekris2-1 -- Accepting UNAUTHENTICATED call from 195.66.73.122: requested format = alaw, requested prefs = (alaw|gsm), actual format = alaw, host prefs = (alaw|gsm), priority = mine -- Executing Set(IAX2/soekris2-2, CALLERID(number)=0327557571) in new stack -- Executing Dial(IAX2/soekris2-2, Zap/2/327557672||tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 2/327557672 Apr 24 22:49:43 WARNING[10273]: chan_iax2.c:7551 socket_read: Received mini frame before first full voice frame -- Zap/6-1 is ringing -- Zap/2-1 is proceeding passing it to IAX2/soekris2-2 -- Zap/1-1 is ringing -- Accepting UNAUTHENTICATED call from 195.66.73.122: requested format = alaw, requested prefs = (alaw|gsm), actual format = alaw, host prefs = (alaw|gsm), priority = mine -- Executing Set(IAX2/soekris2-3, CALLERID(number)=0327557572) in new stack -- Executing Dial(IAX2/soekris2-3, Zap/3/327557673||tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 3/327557673 -- Zap/3-1 is proceeding passing it to IAX2/soekris2-3 -- Zap/2-1 is ringing -- Accepting UNAUTHENTICATED call from 195.66.73.122: requested format = alaw, requested prefs = (alaw|gsm), actual format = alaw, host prefs = (alaw|gsm), priority = mine -- Executing Set(IAX2/soekris2-4, CALLERID(number)=0327557573) in new stack -- Executing Dial(IAX2/soekris2-4, Zap/4/327557674||tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 4/327557674 !! Not good - head of queue has not been transmitted yet -- Accepting UNAUTHENTICATED call from 195.66.73.122: requested format = alaw, requested prefs = (alaw|gsm), actual format = alaw, host prefs = (alaw|gsm), priority = mine -- Executing Set(IAX2/soekris2-5, CALLERID(number)=0327557575) in new stack -- Executing Dial(IAX2/soekris2-5, Zap/5/327557674||tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 5/327557674 -- Hungup 'Zap/3-1' == Spawn extension (biuro, 327557572, 2) exited non-zero on 'IAX2/soekris2-3' -- Hungup 'IAX2/soekris2-3' -- Hungup 'Zap/2-1' == Spawn extension (biuro, 327557571, 2) exited non-zero on 'IAX2/soekris2-2' == Primary D-Channel on span 1 up -- Channel 0/1, span 1 got hangup request == Primary D-Channel on span 1 up -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(IAX2/soekris2-1, ) in new stack ==
[Asterisk-Users] PRI got event: HDLC Bad FCS (8) on Primary D-channel of span
Hello,I get an Error every minute on the second card of two installed TE410P Cards in our System.The error is: PRI got event. HDLC Abort (6) on Primary D-channel of span 5(-8)PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 5(-8)Is it possible that there are known problems with 2 cards in one system?I'm running Asterisk/Libpri/zaptel from SVN branch-1.2-16008I was running Debian Stable with Kernel 2.4.25Since Yesterday i'm running Kernel 2.6.8The Interrupte of the cards are: 16 and 28Do anybody have any idea how i can solve this Problem? -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Background asynchronous AGI
I have been writing a lot of AGI programs in C with good success. I would like somehow to have an AGI program continue in the background while the pbx execution returns to the dialplan and continues. Is this possible? I was thinking that perhaps I could fork or create another thread within the AGI prog. The reason I want to do so is in order to monitor external information (e.g. credit limit and realtime cost of the current call) and then perhaps hang up the call, transfer it or play an announcement to it. I'm aware I could do this with a separate control program using the Manager API, but I like the idea of it being done per-call on demand using AGI if possible. Can anyone suggest any ideas or better techniques? Thanks in advance! Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] annoying noise on analog phones on tdm400p
hmm.. does really nobody had such an issue before? Thomas Artner wrote: Hi! I am using asterisk with two tdm400p cards. Sometimes (one call out of ten), when a call comes in and is taken, there is some terrible noise for a short time in the line (for about a second). Both partys can hear the noise. And sometimes the call has to be hung up, because the noise doesn't disappear. Has anyone any idea where the problem could be? cheers, tom ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)
Am Tuesday 25 April 2006 11:24 schrieb Olle E Johansson: 25 apr 2006 kl. 00.24 skrev Thomas Winter: Am Monday 24 April 2006 18:39 schrieb Doug Lytle: Thomas Winter wrote: Hi, I dont want to have in the SIP HEADER the CALLERID(name) (the Display Name) for the initial INVITE to an SIP proxy. If I use SET(CALLERID(name)=) the display-name is asterisk. Just a guess, try: SET(CALLERID(name)= ) Hi, Asterisk will use this space. - FROM: sip:CALLERID(number)@domain.tld We do insert asterisk when we have no caller ID name. In what situations don't you want a caller ID name at all? I am a bit curious here, in order to understand. Regards /Olle Hi, I have an gateway provider. He accepts mynumber in the displayname: FROM: mynumber sip:[EMAIL PROTECTED] or not using the displayname and number at all: FROM: sip:[EMAIL PROTECTED] if I do not want to show mynumber on the called POTS phone. With * I have allways an displayname in the FROM field and can not disable showing my CALLERID to the called phone. best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Background asynchronous AGI
Hi Tony I have the same problem you have, i think what would you like to do (as me), is to update in a realtime basis credit for prepaid customer, look what I posted today, its from ramcluster and the threat is billing realtime, this is what i discover right now. Hope it help you 2006/4/25, Tony Mountifield [EMAIL PROTECTED]: I have been writing a lot of AGI programs in C with good success. I would like somehow to have an AGI program continue in the background while the pbx execution returns to the dialplan and continues. Is this possible? I was thinking that perhaps I could fork or create another thread within the AGI prog. The reason I want to do so is in order to monitor external information (e.g. credit limit and realtime cost of the current call) and then perhaps hang up the call, transfer it or play an announcement to it. I'm aware I could do this with a separate control program using the Manager API, but I like the idea of it being done per-call on demand using AGI if possible. Can anyone suggest any ideas or better techniques? Thanks in advance! Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another undefined pri_restart failure
Hi: I upgraded SuSE to 10 and Asterisk to trunk and now after deleting all modules and previously compiled stuff and recompiling asterisk, zaptel, and libpri, I get this failure of asterisk to start: [pbx_realtime.so]Apr 25 03:36:41 WARNING[8269]: loader.c:726 __load_resource: new style pbx_realtime.so (0x31) loaded RTLD_LOCAL = (Realtime Switch) [chan_mgcp.so]Apr 25 03:36:41 WARNING[8269]: loader.c:726 __load_resource: new style chan_mgcp.so (0x1) loaded RTLD_LOCAL = (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Found == MGCP Listening on 0.0.0.0:2727 == Using TOS bits 0 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) [chan_zap.so]Apr 25 03:36:41 WARNING[8269]: loader.c:718 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_restart Apr 25 03:36:41 WARNING[8269]: loader.c:850 print_and_load: Loading module chan_zap.so failed! I modified modules.conf to add noload = res_snmp.so, because it fails. I've tried recompiling libpri and everything and modifying path variables. Please help!! ___ For the record, if it is of help Env is: LESSKEY=/etc/lesskey.bin NNTPSERVER=news INFODIR=/usr/local/info:/usr/share/info:/usr/info MANPATH=/usr/share/man:/usr/local/man:/usr/X11R6/man:/opt/gnome/share/man KDE_MULTIHEAD=false SSH_AGENT_PID=6720 HOSTNAME=ScottSuSE DM_CONTROL=/var/run/xdmctl GNOME2_PATH=/usr/local:/opt/gnome:/usr XKEYSYMDB=/usr/X11R6/lib/X11/XKeysymDB GPG_AGENT_INFO=/tmp/gpg-NIZ0pv/S.gpg-agent:17362:1 HOST=ScottSuSE TERM=xterm SHELL=/bin/bash PROFILEREAD=true HISTSIZE=1000 XDM_MANAGED=/var/run/xdmctl/xdmctl-:1,maysd,mayfn,sched,rsvd,method=classic GTK2_RC_FILES=/etc/opt/gnome/gtk-2.0/gtkrc:/opt/gnome/share/themes//Qt/gtk-2.0/gtkrc:/root/.gtkrc-2.0-qtengine:/root/.kde/share/config/gtkrc-2.0 GTK_RC_FILES=/etc/opt/gnome/gtk/gtkrc:/root/.gtkrc:/root/.kde/share/config/gtkrc GNOME_PATH=:/opt/gnome:/usr GS_LIB=/root/.fonts WINDOWID=46137351 OLDPWD=/etc/asterisk QTDIR=/usr/lib/qt3 XSESSION_IS_UP=yes KDE_FULL_SESSION=true GROFF_NO_SGR=yes JRE_HOME=/usr/lib/jvm/java/jre USER=root LS_COLORS=no=00:fi=00:di=01;34:ln=00;36:pi=40;33:so=01;35:do=01;35:bd=40;33;01:cd=40;33;01:or=40;31:ex=00;32:*.cmd=00;32:*.exe=01;32:*.com=01;32:*.bat=01;32:*.btm=01;32:*.dll=01;32:*.tar=00;31:*.tbz=00;31:*.tgz=00;31:*.rpm=00;31:*.deb=00;31:*.arj=00;31:*.taz=00;31:*.lzh=00;31:*.zip=00;31:*.zoo=00;31:*.z=00;31:*.Z=00;31:*.gz=00;31:*.bz2=00;31:*.tb2=00;31:*.tz2=00;31:*.tbz2=00;31:*.avi=01;35:*.bmp=01;35:*.fli=01;35:*.gif=01;35:*.jpg=01;35:*.jpeg=01;35:*.mng=01;35:*.mov=01;35:*.mpg=01;35:*.pcx=01;35:*.pbm=01;35:*.pgm=01;35:*.png=01;35:*.ppm=01;35:*.tga=01;35:*.tif=01;35:*.xbm=01;35:*.xpm=01;35:*.dl=01;35:*.gl=01;35:*.wmv=01;35:*.aiff=00;32:*.au=00;32:*.mid=00;32:*.mp3=00;32:*.ogg=00;32:*.voc=00;32:*.wav=00;32: DESKTOP_LAUNCH=kde-open OPENWINHOME=/usr/openwin XNLSPATH=/usr/X11R6/lib/X11/nls SSH_AUTH_SOCK=/tmp/ssh-IWHyx6676/agent.6676 HOSTTYPE=x86_64 SESSION_MANAGER=local/ScottSuSE:/tmp/.ICE-unix/6784 FROM_HEADER= PAGER=less XDG_CONFIG_DIRS=/usr/local/etc/xdg/:/etc/xdg/:/etc/opt/gnome/xdg/ LD_HWCAP_MASK=0x2000 KONSOLE_DCOP=DCOPRef(konsole-6808,konsole) MINICOM=-c on GNOMEDIR=/opt/gnome DESKTOP_SESSION=default PATH=/sbin:/usr/sbin:/usr/local/sbin:/opt/kde3/sbin:/opt/gnome/sbin:/root/bin:/usr/local/bin:/usr/bin:/usr/X11R6/bin:/bin:/usr/games:/opt/gnome/bin:/opt/kde3/bin:/usr/lib/mit/bin:/usr/lib/mit/sbin CPU=x86_64 JAVA_BINDIR=/usr/lib/jvm/java/bin KONSOLE_DCOP_SESSION=DCOPRef(konsole-6808,session-1) INPUTRC=/etc/inputrc PWD=/usr/src/asterisk/libpri [EMAIL PROTECTED] JAVA_HOME=/usr/lib/jvm/java LANG=POSIX PYTHONSTARTUP=/etc/pythonstart SDK_HOME=/usr/lib/jvm/java SSH_ASKPASS=/usr/lib64/ssh/x11-ssh-askpass TEXINPUTS=::/root/.TeX:/usr/share/doc/.TeX:/usr/doc/.TeX:/root/.TeX:/usr/share/doc/.TeX:/usr/doc/.TeX JDK_HOME=/usr/lib/jvm/java SHLVL=2 HOME=/root LESS_ADVANCED_PREPROCESSOR=no OSTYPE=linux LS_OPTIONS=-a -N --color=tty -T 0 XCURSOR_THEME=crystalwhite WINDOWMANAGER=/usr/bin/dbus-launch --sh-syntax --exit-with-session /usr/X11R6/bin/kde GTK_PATH=/usr/local/lib/gtk-2.0:/opt/gnome/lib/gtk-2.0:/usr/lib/gtk-2.0 LESS=-M -I MACHTYPE=x86_64-suse-linux LOGNAME=root GTK_PATH64=/usr/local/lib64/gtk-2.0:/opt/gnome/lib64/gtk-2.0:/usr/lib64/gtk-2.0 CVS_RSH=ssh XDG_DATA_DIRS=/usr/local/share/:/usr/share/:/etc/opt/kde3/share/:/opt/kde3/share/:/opt/gnome/share/ ACLOCAL_FLAGS=-I /opt/gnome/share/aclocal LC_CTYPE=en_US.UTF-8 DBUS_SESSION_BUS_ADDRESS=unix:abstract=/tmp/dbus-z1RTWWV1Gq,guid=3beb4d44c3081877355afd4083cca800 PKG_CONFIG_PATH=/usr/local/lib/pkgconfig:/usr/local/share/pkgconfig:/usr/lib64/pkgconfig:/usr/share/pkgconfig:/opt/kde3/lib64/pkgconfig:/opt/gnome/lib64/pkgconfig:/opt/gnome/lib64/pkgconfig:/opt/gnome/share/pkgconfig LESSOPEN=lessopen.sh %s USE_FAM= INFOPATH=/usr/local/info:/usr/share/info:/usr/info:/opt/gnome/share/info DISPLAY=:1 XAUTHLOCALHOSTNAME=ScottSuSE LESSCLOSE=lessclose.sh %s
[Asterisk-Users] No sound in one calling direction, men using PRI with E1 and Q.SIG
I've been trying lots of configurations now. And the problem that I can't solve is this: I have a Digium T205P card. I have connected one of the connections to our internal PBX (NEC 2000 IPS). The Asterisk is configured as pri_cpe, and the NEC is configured to be the network side of the connection. Both ends are using b-channels 1-15 and 17-31, the d-channel is on 16. When I start everything, the link is ok on both ends, and it says that the D-channel is up (on both ends). Now, when I try to dial from our internal PBX to the Asterisk, the call connects ok, but there is no sound. But when I dial from the Asterisk to our NEC PBX everything works just fine, and the sound is working perfectly. One more strange thing is that when I dial from the NEC to Asterisk, every time a new B-channel is connecting the call (which I guess is normal). But it NEVER uses channel 31, it skips from 30 to 1. But then it seems to try to connect the call on channel 16, which is the D-channel(!), and that of course fails. Both ends seem to be setup correctly, and since the D-channel is initialized correctly, both ends must be using the correct channel for this, but still Asterisk tries to connect the incomming call B-channel on channel 1-30, instead of 1-15,17-31. Could this be caused by something in the Q.SIG protocol? I have used the NEC PBX with other PBX's, using Q.SIG. And everything has been working just fine. I've tried to look at the PRI debug output, but not much help there... What information is needed from me to get any help with this? Best regards, Peter Olsson Visionutveckling AB ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival , Cannot hear the words after ,
Hi I am trying to use festivall with asterisk , I am using RHEL4 , asterisk1.2.7.1 and festival-1.95-beta , I am able to hear the voice form the text file , when I dial to the extension, but when I have , in my text file , it plays only the text upto , and in the CLI , the , is shown as | I had cut and pasted CLI messages for reference -- Executing Answer(SIP/326-78c7, ) in new stack -- Executing Festival(SIP/326-78c7, Hello | This is Joseph | How are U ) in new stack == Parsing '/etc/asterisk/festival.conf': Found -- Executing Hangup(SIP/326-78c7, ) in new stack == Spawn extension (from-internal, 555, 3) exited non-zero on 'SIP/326-78c7' -- Executing Macro(SIP/326-78c7, hangupcall) in new stack I had followed the link http://www.voip-info.org/wiki/view/Asterisk+festival+installation for the installation Thanks Joseph John Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SQL update failing/long fullcontact
Hi
We have some users who are supplying
very long, broken contact details (from Cisco 7912 phones):
Apr 25 11:29:46 WARNING[1480]
chan_sip.c: No closing bracket found in '1st Floor Scanner - 137 sip:[EMAIL PROTECTED]:5060;user=phone;transport='
Apr 25 11:29:46 NOTICE[1480]
chan_sip.c: '1st Floor Scanner - 137 sip:[EMAIL PROTECTED]:5060;user=phone;transport='
is not a valid SIP contact (missing sip:) trying to use anyway
Any ideas how to stop this? Most of
the time it's harmless but some make the SQL queries so long they overflows
sql in res_config.c:
static struct ast_variable *realtime_mysql(..)
{
char sql[256];
..
snprintf(sql, sizeof(sql), SELECT
* FROM %s WHERE %s%s '%s', table, newparam, op, newval);
..
}
then:
Apr 25 11:29:46 DEBUG[1480] res_config_mysql.c:
MySQL RealTime: Update SQL: UPDATE sip SET ipaddr = 'yyy.yy.yyy.yyy', port
= '25766', regseconds = '1145963986', username = '1st Floor Scanner - 137
sip:', fullcontact = '1st Floor Scanner - 137 sip:[EMAIL PROTECTED]:5060;user=phone;transport='
WHERE name = '84410662
Apr 25 11:29:46 DEBUG[1480] res_config_mysql.c:
MySQL RealTime: Query Failed because: You have an error in your SQL syntax;
check the manual that corresponds
to your MySQL server version for
the right syntax to use near ''84410662' at line 1
The query is 257 bytes so the last quote
is truncated and the update fails.
Should I submit a patch? If nothing
else it'd be nice to check that the query fits into sql and complain if
it doesn't.
Cheers,
Mark Drayton
This message and any attachment are confidential and may be privileged or
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recipient, please telephone or email the sender and delete this message
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Re: [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)
Mike, As someone else mentioned, the delay in getting the disconnect from the CO is a function of the CO equipment and there isn't much you can do about that. In one of my test cases from yesterday, disconnect came within three seconds of the pstn phone hanging up. I'd have to guess that some CO switches probably have some form of timeout parameter that is applied to the entire switch, and the parameter probably has something to do with limiting internal switch issues, conflicts with flash, etc, etc. I'd also guess the delay can probably be traced to specific CO switch vendors, model of switch, etc. In the old electro-mechanical switches, disconnect would happen within a second or two. If a pstn caller listens to someone's entire voicemail greeting and then hangs up, you're going to be stuck with an empty voicemail of whatever duration that you have maxsilence set to in voicemail.conf. Don't think there is anything you can actually do about that. Rich Mike Garey wrote: well, the problem isn't that the card doesn't detect a disconnect, it's that it doesn't detect it immediately (or at least within a short period). I'm talking about 10 or so seconds before the channel is hung up - which is causing empty voicemail messages to be left when the user hangs up before the voicemail starts to record (since the channel sticks around, and asterisk thinks the person is still there). I tried enabling busydetect=yes in zapata.conf, but it didn't make a difference. Mike On 4/24/06, Mark Phillips [EMAIL PROTECTED] wrote: Likewise here. Using a 10 port FXO card and no problems detecting remote hangup. I'll grant you it can be a little slow sometimes however. On Mon, 2006-04-24 at 16:54 -0500, Rich Adamson wrote: Mike Garey wrote: As far as I can tell, after discussing this matter with other asterisk users in my area, my telco _does_ provide disconnect supervision.. It seems that the problem is actually related to the Sangoma A200 card I'm using, as two other people both using this same card have expressed the same problem.. Are there any other users on this list using the Sangoma A200 FXO port card, and experiencing problems with asterisk not detecting when a channel has been disconnected? Thanks, Hasn't been a problem here with either the TDM400 or A200D cards (both are in use in same box). Just tested it again from an external pstn phone, calling into asterisk. When the pstn phone hangs up, asterisk recognized it and dropped the sip session that was handling the call (to a 7960). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] MeetAsterisk in Europe - register today!
Hi Olle, Very well, but can we do for you during the french day in Paris and what are the conditions ? I have announced your event near to all of our resellers/fitters in our country. I have talked about that event until our African contacts ;-) Best Regards, Francois BERGERET. http://www.ges.fr/bin/[EMAIL PROTECTED] GES, 205, rue de l'Industrie B.P.46 77542 SAVIGNY-LE-TEMPLE Cedex France VAT-ID FR 53 787 350 016 Tel : +33 1 64.41.78.88 Fax : +33 1 60.63.24.85 VoIP IAX2 : [EMAIL PROTECTED] -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Olle E Johansson Envoyé : vendredi 21 avril 2006 04:46 À : Asterisk Non-Commercial Discussion Users Mailing List - Objet : [Asterisk-Users] MeetAsterisk in Europe - register today! Friends, Beginning next week, I will travel around Europe to teach Asterisk - the one day Meet Asterisk training. MeetAsterisk is organized by Edvina in cooperation with Digium and Voop. In many places, local Asterisk equipment resellers participate and show their equipment. This is the tour plan: * Amsterdam April 26 * Copenhagen April 27 * Oslo April 28 * Paris May 3 * Brussels May 4 * London May 5 * Stockholm May 19 (Close to Von Europe) MeetAsterisk is the one-day training that introduces Asterisk for a beginner, both from a business perspective and a technical perspective. You will get insights in how to use Asterisk in your business, as well as an introduction in how to install and set up Asterisk. It's a day filled with information to give you a quick-start with Asterisk. Find out the complete schedule at http://www.meetasterisk.com and register today! See you at MeetAsterisk! /Olle PS. MeetAsterisk will also contain a brief introduction to the new functions in the coming version of Asterisk - Asterisk 1.4 - to be released this summer. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMP kernel on Pent 4?
Mike Fedyk wrote: Rich Adamson wrote: Had a Pent 4 server running fc3 crash (kernel panic) and am rebuilding from scratch. I installed FreePBX (CentOs) from scratch and asterisk was running, but had not yet been configured. It too crashed with a kernel panic. Ran memtest for 24 hours; no errors or issues uncovered. I then noticed that FreePBX installed using a SMP kernel (and grub indicated a non-SMP kernel was installed as well). Would running an SMP kernel on a Pent 4 potentially cause a kernel panic? (Or, do I need to dig somewhere else?) Nothing in the logs to suggest a root cause and I'm now waiting on recurrence using the non-SMP kernel. Were you able to see an oops message when it crashed? If not, then make sure a X11 server isn't running, and turn on nmi_watchdog. No. In the FreePbx default installation, CentOs and all of the asterisk components are installed automatically. X11 is not installed, leaving only a linux command line on the console. Since the screen has only 24 displayable lines, the interesting stuff scrolled off the top before the kernel panic occurred. The easiest way to capture the oops is with a serial console, but hand typing the text into another computer or a snapshot has worked in the past also. Then post your results. Also check the system temp with lm_sensors and the quality of your drives with smartctl. I'll give those a try. Gut feeling is oriented around FreePbx defaulting to an smp kernel and this particular system is a single-processor single-core Pent 4. I changed grub to load a non-smp kernel and still waiting on recurrence (after about 24 hours). R. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] res_perl voor asterisk 1.2.4
Title: Running commands from dialplans Hi, Can anybody tell me which version of res_perl I have to install on Asterisk 1.2.4. I tried to compile res_perl version 3.5 on Asterisk 1.2.4 and I got the following error. gcc -Wall -DRES_PERL_BASE=\/usr/local/res_perl\ -DMULTIPLICITY - D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE - D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm - I/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE - I/usr/src/bristuff-0.3.0-PRE-1l/asterisk-1.2.4/ - I/usr/src/bristuff-0.3.0-PRE-1l/asterisk-1.2.4//include -I. -c AstAPIBase.c AstAPIBase.c: In function `asterisk_recordfile': AstAPIBase.c:435: warning: ISO C90 forbids mixed declarations and code AstAPIBase.c: In function `asterisk_request_and_dial': AstAPIBase.c:813: warning: passing arg 6 of `ast_request_and_dial' makes integer from pointer without a cast AstAPIBase.c:813: error: too few arguments to function `ast_request_and_dial' AstAPIBase.c: In function `asterisk_request': AstAPIBase.c:880: error: too few arguments to function `ast_request' make: *** [AstAPIBase.o] Error 1 Can anybody tell me if this version is the right res_perl version? Kind regards. Arjan Kroon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail being cut-off
Hi, I have 2 installs complaining of the same problem. They are both using Asterisk 1.0.10. They complain that when someone leaves a message, they are being cut-off. We tried playing with the maxsilence, silencethreshold and maxmessage without sucess. Any hints? Thanks, Andre Courchesne ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly (UPDATE)
I think my DTMF problems are solved, but the solution isn't crystal clear. I reverted back to 1.2.6 and then had the idea to have asterisk email me every time someone hit the invalid extension. The email contained the number they dialed and the channel (read sipura box) they came in on. After a day and a half I noticed 99.9% of my invalid extensions were all coming from one sipura box that happened to be running version 3.1.10 firmware. I downgraded to 3.1.3 and the DTMF problems disappeared. Thing is, I *know* I was getting invalid extensions on calls from the other sipura box (which is running 3.1.7) when I was running asterisk 1.2.7.1 but I don't know how many and I'm not going back to find out just yet. (Need to wait for my users to stop thinking its a useless POS first). So there *may* still be an issue with 1.2.7.1, but I definitely had an issue with sipura firmware 3.1.10 and DTMF detection. Thanks to everyone who submitted ideas. -Dave Dave Fullerton wrote: I have reverted back to 1.2.6 and set my sipuras to tx dtmf as info so I can see them with sip debug. I'll see if there is a difference and report on my findings in a couple days. -Dave Bryan Boatright wrote: I too am experiencing DTMF problems with 1.2.7.1 that I did not experience with recent prior versions. I've backed up to version 1.2.6 and so far DTMF detection is working reliably (but that's only with about 10 calls worth of testing). I've only had problems over SIP channels. Zap channels did not have problems with 1.2.7.1. I do not have any IAX channels, so cannot comment on that. I know others tend to discount DTMF problems because of known problems with how Asterisk handles DTMF, but there does seem to be enough anecdotal evidence that something bad has recently happened to make things worse. Dave, would you mind trying version 1.2.6 to see if that also resolves your problems? Dave Fullerton wrote: Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) - Sipura 3k (fxo) - Asterisk -IAX- remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto attendant (I don't have any FXO ports available on our PBX to do it the right way). The attendant answers and asks them the remote extension they want to dial. This setup has worked very well for several months. Last week I upgraded to 1.2.7.1 from 1.2.4 (I think). Since then I've been having trouble with the auto-attendant correctly detecting DTMF (missing digits). Some times it works flawlessly, others I have to try over and over before it is detected correctly. It isn't even consistently dropping the same digit from what I can see on the console. The only thing I've found is that I have a better chance of it working if I wait for the prompt to finish before dialing. I have changed the DTMF method from rfc2833 to info and finally inband with only a little change (inband seems to work the best). Has anyone else run into similar problems or have any more suggestions to try? This is the attendant portion of my extensions.conf: [inter-attendant] exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Set(TIMEOUT(response)=10) exten = s,4,Background(enter-ext-of-person) exten = i,1,Playback(invalid) exten = i,2,Goto(s,4) exten = i,3,Hangup exten = t,1,Playback(goodbye) exten = t,2,Hangup include = tests include = fullertonpbx include = intercompany Thank you for any insight you can provide. Dave Fullerton ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)
Olle E Johansson wrote: 25 apr 2006 kl. 00.24 skrev Thomas Winter: Am Monday 24 April 2006 18:39 schrieb Doug Lytle: Thomas Winter wrote: Hi, I dont want to have in the SIP HEADER the CALLERID(name) (the Display Name) for the initial INVITE to an SIP proxy. If I use SET(CALLERID(name)=) the display-name is asterisk. Just a guess, try: SET(CALLERID(name)= ) Hi, Asterisk will use this space. - FROM: sip:CALLERID(number)@domain.tld We do insert asterisk when we have no caller ID name. In what situations don't you want a caller ID name at all? I am a bit curious here, in order to understand. It would be nice to NOT have that feature Is there an easy way to disable that? John Novack Regards /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lastest stable build
Hi, What is the version number of the lastest stable release, and how to get it through CVS or wget? Thnx. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lastest stable build
25 apr 2006 kl. 15.34 skrev Wai Wu: Hi, What is the version number of the lastest stable release, and how to get it through CVS or wget? Thnx. All of the information you look for is easily available on http://www.asterisk.org /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Really Old Rotary Phone
Ok... I am not a telephone guy... I was born after rotary phones, so forgive my ignorance in this matter. I am trying to get a really old rotary phone up and running with an ATA. Why? Who knows... just thought it would be cool. The problem is that it does not have an RJ11 connector, instead it has three wires (green,yellow,red). Does anyone know what that type of connector is called? Or know of a reference to build an adapter to 2 line? Thanks, Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Really Old Rotary Phone
Yellow=ground - not used Green = tip Red = ring connect green/red to rj pins 4/5 You could pick up a quarter mod line cord (mod to spade) and replace the cord, or use a screw terminal block to connect to line. Enjoy On Apr 25, 2006, at 9:19 AM, Sean Cook wrote: Ok... I am not a telephone guy... I was born after rotary phones, so forgive my ignorance in this matter. I am trying to get a really old rotary phone up and running with an ATA. Why? Who knows... just thought it would be cool. The problem is that it does not have an RJ11 connector, instead it has three wires (green,yellow,red). Does anyone know what that type of connector is called? Or know of a reference to build an adapter to 2 line? Thanks, Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Really Old Rotary Phone
Red and Green are the Tip and Ring. Yellow may have to be strapped to one or the other depending on the phone you have. Some phones may not ring at all, due to special frequency ringers installed in them for party lines. Western Electric did not use these. As to the ATA, MOST ATA's do not support pulse dial, though the IAXy does. Asterisk does in the TDM400, though the decoding code requires some modification for dial speed variance. There are a group of telephone switch collectors that use Asterisk as a tandem switch to interconnect old ( very old to you ) switches through the Internet, so what you want to do really isn't that strange. John Novack Sean Cook wrote: Ok... I am not a telephone guy... I was born after rotary phones, so forgive my ignorance in this matter. I am trying to get a really old rotary phone up and running with an ATA. Why? Who knows... just thought it would be cool. The problem is that it does not have an RJ11 connector, instead it has three wires (green,yellow,red). Does anyone know what that type of connector is called? Or know of a reference to build an adapter to 2 line? Thanks, Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Two asterisk process in one hardware.
Hello I'm using voicemail with realitime. And I need use two diferent and separate databases. thanks. jsalas -Mensaje original- De: Mike Fedyk [mailto:[EMAIL PROTECTED] Enviado el: Monday, April 24, 2006 8:24 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Two asterisk process in one hardware. Juan Salas wrote: Hello. Has anybody knows how run two asterisk process in one hardware? (each one with its own configuration?) What end outcome do you want? Maybe there is another way to do it... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Some questions re. T1 cards QoS
hugolivude wrote: Funny you mention that Kevin. I was on the web site this morning and I saw it here: http://www.digium.com/en/products/hardware/analogcards.php Later on the same day, that page had changed. The text was gone and the TDM2400P TDM400P had swapped positions... I'll mention it to our web team, thanks. So a 4 span, is 4 T1 lines (wow). With a single span, I'd set echocancellation=yes or similar in zapata.conf? Actually, you would do that regardless, unless you don't want echo cancellation at all. If the hardware echo canceler is available and not disabled manually, it will be used instead of the software version; if not, the software canceler will be used. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] About Softphone IAX free for Pocket PC
Hello, Has anyone Knowledge about softphone IAX for pocket PC totally free? Tkanks for all. -- Sandra Salmerón Ntutumu[EMAIL PROTECTED] Tlf. Analog: +34 914888405 / Móvil: 653574298 Tlf. IP desde FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010 Fundación EHAS: Enlace Hispanoamericano de Salud - www.ehas.org Telemedicina rural para zonas aisladas de países en desarrollo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Really Old Rotary Phone
Jerry Jones wrote: Yellow=ground - not used Green = tip Red = ring connect green/red to rj pins 4/5 You could pick up a quarter mod line cord (mod to spade) and replace the cord, or use a screw terminal block to connect to line. Enjoy This worked perfectly! Thank you! Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Really Old Rotary Phone
On 4/25/06, Sean Cook [EMAIL PROTECTED] wrote: This worked perfectly! Thank you! Sean Now, I think the question is, does your ATA actually support rotary/pulse dialing? Mine (SPA-2000) did not. I bought a (very cheap) MITEL-1 Smart Dialer and went through a RIDICULOUS amount of pain trying to configure it to convert from pulse to DTMF dialing, and it did sort of work although I never seemed to be able to get it configured exactly the way I wanted it. I ended up getting a TDM400 with a couple of FXS modules (which I had been needing to get anyway), and that worked perfectly after patching the pulse-dial debounce code in Zaptel (although I believe the newest version of Zaptel already comes with the needed changes). If there are ATAs that support pulse dialing, I'd like to know about it, because now I would like to be able to use my pulse phone in locations other than the physical location of my Asterisk machine with the TDM400 card in it. So if you find that yours works, could you please let me know? Thanks, Rusty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] About Softphone IAX free for Pocket PC
Unless you have a top of the line Pocket PC don't even bother. Most inexpensive units like the T-Mobile MDA just dont have the processing power to handle VoIP. I have tried ESJPhone, SJPhone, and some other one which I forgot about already and the sound quality was horrible regardless of using GPRS or WiFi. That would have been a great benefit to me but its just not going to happen on a device that barely runs Windows Mobile as it is. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of makevuy Sent: Tuesday, April 25, 2006 8:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] About Softphone IAX free for Pocket PC Hello, Has anyone Knowledge about softphone IAX for pocket PC totally free? Tkanks for all. -- Sandra Salmerón Ntutumu[EMAIL PROTECTED] Tlf. Analog: +34 914888405 / Móvil: 653574298 Tlf. IP desde FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010 Fundación EHAS: Enlace Hispanoamericano de Salud - www.ehas.org Telemedicina rural para zonas aisladas de países en desarrollo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Really Old Rotary Phone
I do have a TDM400 and the Sangoma A200. I have done pulse with the TDM400, but have not with the A200. I have just never seen a phone like this... ;) Rusty Dekema wrote: On 4/25/06, Sean Cook [EMAIL PROTECTED] wrote: This worked perfectly! Thank you! Sean Now, I think the question is, does your ATA actually support rotary/pulse dialing? Mine (SPA-2000) did not. I bought a (very cheap) MITEL-1 Smart Dialer and went through a RIDICULOUS amount of pain trying to configure it to convert from pulse to DTMF dialing, and it did sort of work although I never seemed to be able to get it configured exactly the way I wanted it. I ended up getting a TDM400 with a couple of FXS modules (which I had been needing to get anyway), and that worked perfectly after patching the pulse-dial debounce code in Zaptel (although I believe the newest version of Zaptel already comes with the needed changes). If there are ATAs that support pulse dialing, I'd like to know about it, because now I would like to be able to use my pulse phone in locations other than the physical location of my Asterisk machine with the TDM400 card in it. So if you find that yours works, could you please let me know? Thanks, Rusty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Updated: No audio when dialing in via PRI with Q.SIG
After lots of testing I discovered that I could get the sound to work. The only thing I had been testing was MeetMe and Voicemail. But when I dialed a SIP-phone, or routed back to other phones via the PRI interface, everything works just great! The problem only seem to occur when dialing directly into Asterisk, when Asterisk sends the audio output. I have also discovered that the PRI never seem to get the signal that the call has been connected when dialing into MeetMe, it thinks it's still in the ringing state - I've discovered this by watching TAPI events showing up on my other PBX. Is this some kinf of known bug in Asterisk? I guess it's because of this I won't get any sound on these calls When dialing to a SIP phone I get all information. If anyone have any idea, I'd appreciate it. If it helps I could also send some debug logs from ISDN. Best regards, Peter Olsson Visionutveckling AB ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Really Old Rotary Phone
Rusty Dekema wrote: On 4/25/06, Sean Cook [EMAIL PROTECTED] wrote: Now, I think the question is, does your ATA actually support rotary/pulse dialing? Mine (SPA-2000) did not. Most/all of the SIP based ones seem not to. I bought a (very cheap) MITEL-1 Smart Dialer and went through a RIDICULOUS amount of pain trying to configure it to convert from pulse to DTMF dialing, and it did sort of work although I never seemed to be able to get it configured exactly the way I wanted it. Several collectors with similar needs have had good results with the SMART-1, though it can be painful, and there are MANY different versions, even for the US market. The Euro one is reported to be somewhat easier. Keep in mind that it was designed as a store and forward dialer, so it works a little oddly in this application. I ended up getting a TDM400 with a couple of FXS modules (which I had been needing to get anyway), and that worked perfectly after patching the pulse-dial debounce code in Zaptel (although I believe the newest version of Zaptel already comes with the needed changes). Dial speed and make-break ratio were also problems with older dials If there are ATAs that support pulse dialing, I'd like to know about it, because now I would like to be able to use my pulse phone in locations other than the physical location of my Asterisk machine with the TDM400 card in it. So if you find that yours works, could you please let me know? Let the list know. There are others with a similar requirement. The IAXy is the only one I have found that supposedly supports pulse dial. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question on connecting to another system
Hi, If I am interfacing with a legacy PBX system a few questions. #1 What do I need to do to configure 1 port on a dual port card as pri_cpe and another as pri_net? Do I just change my config half-way through the zaptel.conf file? #2 When I setup span=1,1,0,esf,b8zs doesn't the esf indicate that I am using 'Robbed Bit', but then in /etc/asterisk/zapata.conf I have switchtype=national, which seems to indicate 'ISDN signaling'. That is the configuration I have to my current CLEC.. so am I using robbed bit... or ISDN signaling? I DO get caller-id on inbound. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Really Old Rotary Phone
Sean Cook wrote: I do have a TDM400 and the Sangoma A200. I have done pulse with the TDM400, but have not with the A200. The A200 works with pulse dial. If yours does not, contact Sangoma or use the latest drivers. They fixed it after I contacted them several weeks ago. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Softphone IAX free for Pocket PC
On 4/25/06, Kerry Garrison [EMAIL PROTECTED] wrote: Unless you have a top of the line Pocket PC don't even bother. Most inexpensive units like the T-Mobile MDA just don't have the processing power to handle VoIP. I have tried ESJPhone, SJPhone, and some other one which I forgot about already and the sound quality was horrible regardless of using GPRS or WiFi. That would have been a great benefit to me but its just not going to happen on a device that barely runs Windows Mobile as it is. Kerry Garrison Unfortunately, I have to agree. I was very pumped about being able to use VoIP over WiFi on the PPC-6700 (which has a 416 MHz cpu), but the phone's processor just didn't seem to be able to keep up with the RTP stream. It was really unusable; I ended up having to cancel the service and return the device, which was a real shame, because the phone and the EV-DO data service worked quite well in general otherwise. -Rusty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Updated: No audio when dialing in via PRI withQ.SIG
Add an Answer() as your first step in your dialplan and see if that help. snip ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] About Softphone IAX free for Pocket PC
Same results here with my PPC-6700 nice phone no processing power, I found that my EVDO card on Laptop works great with SIP softphones. Unfortunately, I have to agree. I was very pumped about being able to use VoIP over WiFi on the PPC-6700 (which has a 416 MHz cpu), but the phone's processor just didn't seem to be able to keep up with the RTP stream. It was really unusable; I ended up having to cancel the service and return the device, which was a real shame, because the phone and the EV-DO data service worked quite well in general otherwise. -Rusty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] About Softphone IAX free for Pocket PC
I use IaxComm with good results on axim x51 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Tuesday, April 25, 2006 11:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] About Softphone IAX free for Pocket PC Unless you have a top of the line Pocket PC don't even bother. Most inexpensive units like the T-Mobile MDA just dont have the processing power to handle VoIP. I have tried ESJPhone, SJPhone, and some other one which I forgot about already and the sound quality was horrible regardless of using GPRS or WiFi. That would have been a great benefit to me but its just not going to happen on a device that barely runs Windows Mobile as it is. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of makevuy Sent: Tuesday, April 25, 2006 8:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] About Softphone IAX free for Pocket PC Hello, Has anyone Knowledge about softphone IAX for pocket PC totally free? Tkanks for all. -- Sandra Salmerón Ntutumu[EMAIL PROTECTED] Tlf. Analog: +34 914888405 / Móvil: 653574298 Tlf. IP desde FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010 Fundación EHAS: Enlace Hispanoamericano de Salud - www.ehas.org Telemedicina rural para zonas aisladas de países en desarrollo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
Also note that the Smart Jack allows the Telco to provide T1 Signalling in places that it couldn't in the past, most smart jacks that I have used are: [CO]-Optical-[Hut DMS]--[Hut Smart Jack]-HDSL-[CPE Smart Jack] For the list, Telco Techs, mostly do as they are told, and are schooled by the Telco vendors. On 4/24/06, Rich Adamson [EMAIL PROTECTED] wrote: Alexander Lopez wrote: I was once told by a lineman that the cables they use didn't have that many twists in them because it wasn't needed, and that the extra twists would effectively use more cable and thus cost and weigh more than triple what they do now. Good thing he doesn't work for a cable manufacturer as that's a total crock of crap that even an inexperienced person should be able to detect. (You can't twist two wires to make them weight three times as much, or cost three times as much.) He told me that with the number of twists in the Cat 5 cable it would cancel out any interference, but he also stated that the effective length was calculated using a cable with less twists and subsequently 'less dense' and that if using a Cat5e cable you must factor that in. so if you use cat5e cable your are fine but you can't go as far. Essentially true, but the impedance of a T1 cable is different from Cat5 cables, which is one of the primary factors in limiting distance. Has nothing to do with the twists. Shielded vs non-shielded has to do with the environment, and how much electrical noise there is near the T1 cable. Nothing more, nothing less. Regarding the Smart Jack it is mostly used as a location at the CPE where the Telco can loop and make sure that the problem is at your end. So your assumption is correct that you can plug anything you want into it, its one your side of the demark, so if it doesn't work it's YOUR problem. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Softphone IAX free for Pocket PC
Robert Augustyn wrote: I use IaxComm with good results on axim x51 Is that something you developed yourself? If so, can you share it? For the last year I have been trying to find time to get iaxcomm working on a WinCE machine. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P: flash on analog phones doesn't work
Hi, I have a TDM400P (31B) in a PIV 2.8, 512Mb ram, CentOS 4.3, zaptel 1.2.5 and Asterisk 1.2.7.1 and a couple of standard analog phones with a flash button. A hook flash works fine for setting up a 3way call. But pressing the flash button doesn't do anything. The zapata config is below. Anyone have an idea what I'm doing wrong? [channels] context=local usercallerid=yes hidecallerid=no immediate=no transfer=yes threewaycalling=yes canpark=yes echocancel=yes busydetect=yes signalling=fxo_ks group=1 callerid=ANALOG1 1001 channel = 1 signalling=fxo_ks group=1 callerid=ANALOG2 1002 channel = 2 signalling=fxo_ks group=1 callerid=ANALOG3 1003 channel = 3 signalling=fxs_ks group=2 channel = 4 Thanks and regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMS to call back
Awhile back I remember someone posted a SMS to DISA AGI script. I searched the archives and found nothing...anyone out there remember? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Background asynchronous AGI
Can't you do all of this with the (Absolute) time setting? So if the person has 4,000 minutes left.. set the call length for 4,000 minutes as the absolute max. Alternately... you could probably use screen? Launch an AGI from the main AGI using screen so it goes into the background... You could also try writing a daemon in perl I suppose. On 4/25/06, random cluster [EMAIL PROTECTED] wrote: Hi Tony I have the same problem you have, i think what would you like to do (as me), is to update in a realtime basis credit for prepaid customer, look what I posted today, its from ramcluster and the threat is billing realtime, this is what i discover right now. Hope it help you 2006/4/25, Tony Mountifield [EMAIL PROTECTED]: I have been writing a lot of AGI programs in C with good success. I would like somehow to have an AGI program continue in the background while the pbx execution returns to the dialplan and continues. Is this possible? I was thinking that perhaps I could fork or create another thread within the AGI prog. The reason I want to do so is in order to monitor external information (e.g. credit limit and realtime cost of the current call) and then perhaps hang up the call, transfer it or play an announcement to it. I'm aware I could do this with a separate control program using the Manager API, but I like the idea of it being done per-call on demand using AGI if possible. Can anyone suggest any ideas or better techniques? Thanks in advance! Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unicall MFC problems in 0.0.3+asterisk 1.2
Hi Steve and everyone I have a very strange problem with an old 1st gen TDM410P card I've been using in the production machine (10-15K calls/day) without problem with Asterisk 1.0.7+Unicall 0.0.2. When I switched to Asterisk 1.2 and Unicall 0.0.3 (even in 1.2.7.1 and pre9), span 3 handled mfc signalling awfully. Spans 1,2 and 4 worked fine. I've tested it in both 2.4 and 2.6 linux kernels in a Debian 3.1 distribution, and in 2 different servers, with 2 different E1 telco lines, with the same results. 2nd gen cards work just fine in every span. This is the log from a couple of incoming calls. The right ANI/DNIS were 2214291350 and 0200. Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 0001 [1/ 1/Idle /Idle ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 Detected Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 Making a new call with CRN 32769 Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 1101 - [2/ 2/Idle /Idle ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: Unicall/79 event Detected Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 0 on [2/ 2/Seize ack /Seize ack] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 on - [2/ 2/Seize ack /Seize ack] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 0 off [2/ 2/Group A /Category req ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 off - [2/ 2/Group A /Category req ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 1 on [2/ 2/Group A /Category req ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 on - [2/ 2/Group A /Category req ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 1 off [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 off - [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 1 on [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 on - [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 1 off [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 off - [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 2 on [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 on - [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 2 off [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 off - [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 2 on [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 on - [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 2 off [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 off - [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 2 on [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 on - [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 2 off [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 off - [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 2 on [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 on - [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 2 off [2/ 2/Group A /ANI request ] Apr 25 11:08:17 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 off - [2/ 2/Group A /ANI request ] Apr 25 11:08:18 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 2 on [2/ 2/Group A /ANI request ] Apr 25 11:08:18 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 on - [2/ 2/Group A /ANI request ] Apr 25 11:08:18 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 - 2 off [2/ 2/Group A /ANI request ] Apr 25 11:08:18 WARNING[18731] chan_unicall.c: MFC/R2 UniCall/79 5 off - [2/ 2/Group A
[Asterisk-Users] MFCR2 in Brazil, someone?
Does anybody have a working Asterisk server with Unicall using MFCR2 in Brazil? Were having problems. It seems SPANDSP never detect the tones from the telco. Im using brazil protocol variant. Im having lots of problems to find out why spandsp seems to not detect the MF tones. We send the first digit, the telco says they receive it, and respond with the proper signal to ask for the next digit, we just never detect the tone and the T1 timer times up. Some custom logs i have put in mfcr2.c point to spandsp r2_mf_rx always returning a zero value, what seems to mean OFF TONE, because it automatically sends the code to mf_tone_off_event() but without expecting tone because it never enters to mf_tone_on_event() something like this: OUR PBX = seize TELCO = seize ACK === == First DNIS tone == here we never detect the tone from the telco the server is Linux switch-cwb.jeffnetworks.com 2.6.9-34.ELsmp #1 SMP Thu Mar 9 06:23:23 GMT 2006 x86_64 x86_64 x86_64 GNU/Linux already tried different spandsp versions without success. Thanks in advance. -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] About Softphone IAX free for Pocket PC
Steve, I a sorry, I should have verified what I am writing. The software is PPCIAX2 and you can find it: http://www.voipalia.com/ppciax/ Is it pretty not but it works. robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Tuesday, April 25, 2006 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] About Softphone IAX free for Pocket PC Robert Augustyn wrote: I use IaxComm with good results on axim x51 Is that something you developed yourself? If so, can you share it? For the last year I have been trying to find time to get iaxcomm working on a WinCE machine. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Updated: No audio when dialing in via PRIwithQ.SIG
I've already tried that, but the result is the same... :( I've also seen the same error reported a long time ago, on this link: http://lists.digium.com/pipermail/asterisk-users/2004-August/053365.html. But I can't find a solution anywhere... Best regards, Peter Olsson Visionutveckling AB Från: [EMAIL PROTECTED] genom Alexander Lopez Skickat: ti 2006-04-25 18:07 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: RE: [Asterisk-Users] Updated: No audio when dialing in via PRIwithQ.SIG Add an Answer() as your first step in your dialplan and see if that help. snip ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:444e4a53124942303717053! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto Logout from queue
i have a client that wants a function that will automatically logout an agent from a queue if they do not answer a call. This would prevent future calls from being sent to that phone if the agent forgot to logout. Any ideas? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Really Old Rotary Phone
Well it works! The pulse detection is a little squirrelly, even with the debounce changes to wctdm.c. I can't get an audible ring but it does work. Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto Logout from queue
Use the local channel to call the agent first, and if there is no answer, log them out. From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Tue 4/25/2006 2:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Auto Logout from queue i have a client that wants a function that will automatically logout an agent from a queue if they do not answer a call. This would prevent future calls from being sent to that phone if the agent forgot to logout. Any ideas? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.techdatapros.com http://www.techdatapros.com/ winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 21, Issue 132
Hi All I want tosetting as belows.caller --- call ( from telco) -- asterisk --- IVR -- SIP 1. after that, SIP1 transfer to SIP2 (unattendant or attendant transfer). i want to SIP1 hear stream sound data of call conversation between caller and SIP 2 (don't used call conference) SIP3 want to hear stream sound data ofcaller and SIP2 conversation, can be press DTMF keys as: form example: *8401 ( 401 as username of SIP2). could you like to help me tosetup that function. Best regards__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip t38 gateway tests
Hello, I patched asterisk patched with the latest t38 support . I would need some people for tests. Regards harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Restart and Dropped calls
Issam, Don't mean to press, but did you have a solution or a similar experience? - Original Message - Date: Tue, 25 Apr 2006 01:37:18 +0100 From: issam [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Channel Restart and Dropped calls To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed; charset=iso-8859-1; reply-type=original - Original Message - From: Chris Gamble [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, April 24, 2006 9:01 PM Subject: [Asterisk-Users] Channel Restart and Dropped calls We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are getting frequent restarts on the spans which lead to dropped calls. I have pasted some hopefully pertinent information below -- anyone have any clues that might help? Thanks Next line is repeated throughout messages, going through every channel in every connected span. asterisk/full.1:Apr 24 01:15:25 VERBOSE[4196] logger.c: -- B-channel 0/1 successfully restarted on span 1 lspci -v 06:03.0 Communication controller: Digium, Inc. Wildcard TE405P (2nd Gen) (rev 02) Flags: bus master, medium devsel, latency 32, IRQ 201 Memory at 30004a00 (32-bit, non-prefetchable) [size=128] /proc/zapatel/1-4 with 5 being ZTDUMMY Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 B8ZS/ESF ClockSource Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 B8ZS/ESF RED Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help on chan_misdn and MSN's
Quick question: Is there a way to distinguish between calling MSN's when using chan_misdn? More info: I've got my ISDN2 (EuroISDN) up and running here in Romania with 1 base number plus 5 MSN's. Now I want to my * to do different things when receiving a call on from different MSN's (like forwarding the call to my FAX machine or forwarding the call to my mobile). The obvious way of doing this would be to set up different sections in the misdn.conf file for the same port (I only have one port), using different settings for the msns. Unfortunately it seems that the channel driver will only remember the last section it sees for a given channel so I can only use * as the msn - and that defeats the purpose. If any other info is required I'll happily provide it. I'm not including any other info at the moment because I'm unable to filter the list myself and the list of things I've been doing today is very long (starts with downloading kernel 2.6.16.11 off kernel.org, patching for mISDN, downloading chan_misdn, compiling everything... waaay too long list, most of it irrelevant) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Really Old Rotary Phone
Sean Cook wrote: Well it works! The pulse detection is a little squirrelly, even with the debounce changes to wctdm.c. I can't get an audible ring but it does work. Sean By audible ring do you mean you can't get the phone to ring? If that is the case, and you have tried connecting the yellow wire to Green, then to Red, then more information is needed about the phone itself. Contrary to information posted earlier, the Yellow wire often is needed, and was only connected to ground in 2 party service ( remember party lines? ) Contact me off list if need be. If you mean audible ringback, that is an Asterisk issue, John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FastAGI Connection Failure and Hangup
Does anyone know how to make fastagi continue to the next priority if it can not connect to the remote AGI Server? Right now I am just getting Hangup and cant find anything on the net about this. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with using Asterisk with PlusNet in the UK
Hello, I hope someone who has been successful in getting Plus.Net's VOIP service to interface with Asterisk might be able to help. For some reason I can't seam to register or make outgoing calls. If anyone would mind posting their register line as well as the Plus.Net context in the sip.conf file that would probably help me figure out what I need to put into my sip.conf. I've seen references in this lists's archives saying that at least a couple of people have it working but they didn't say how. Thanks in advance, Monty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Restart and Dropped calls
On 4/24/06, Chris Gamble [EMAIL PROTECTED] wrote: We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are getting frequent restarts on the spans which lead to dropped calls. I have pasted some hopefully maybe this is related: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf (search for resetinterval) Please feedback if it 'is' related, i'm curious to know if it helps.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Splitting Zap channels into trunks?
On a TDM2400 with 3 FXO modules, is there a way to split each line into basically being its own trunk or another way to pull off the following scenerio: PBX has 12 inbound PSTN lines 1,3,5,7 are the 714 phone number hunt group 2,4,6,8 are the 888 phone number hunt group 9-12 are fax lines Customer wants outbound calls to go out in the following order: 8,7,6,5,4,3,2,1,12,10,11,9 Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One Way Audio....in the middle of a call
We had a user report that they were on a SIP --- PSTN call for about 4.5 minutes before the call went to on-way audio. The user called the person back and they reported being able to hear my user, but my user couldn't hear them. The audio condition persisted for about 15 seconds before the user hung up. Where do I start to troubleshoot one way audio that occurs during a call? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto Logout from queue
Yes, that is the functionality I am looking for, just not sure how exactly to pull that off. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Tuesday, April 25, 2006 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Auto Logout from queue Use the local channel to call the agent first, and if there is no answer, log them out. _ From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Tue 4/25/2006 2:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Auto Logout from queue i have a client that wants a function that will automatically logout an agent from a queue if they do not answer a call. This would prevent future calls from being sent to that phone if the agent forgot to logout. Any ideas? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.techdatapros.com/ http://www.techdatapros.com attachment: winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Way Audio....in the middle of a call
Hi Geoff, You might want to try tcdump, specifying the source and destination IP (to minimize the info) and see where are the RTP packets going ; youwill see if they change port or something like that after a while. Cheers, Frederic - Original Message - From: Geoff Manning To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, April 25, 2006 17:37 Subject: [Asterisk-Users] One Way Audioin the middle of a call We had a user report that they were on a SIP --- PSTN call for about 4.5 minutes before the call went to on-way audio. The user called the person back and they reported being able to hear my user, but my user couldn't hear them. The audio condition persisted for about 15 seconds before the user hung up. Where do I start to troubleshoot one way audio that occurs during a call? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410 and 411
If I remove the eco cancellation module from a TE411P card, will it work as a plain TE410P? -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410 and 411
Carlos Chavez wrote: If I remove the eco cancellation module from a TE411P card, will it work as a plain TE410P? Yes. You can also the 'vpmsupport=0' module parameter to disable the use of the module without physically removing it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queues that do not play music
Is there a way to send a caller directly to the queue member if one is available, without answering it and putting him on MOH? Note that playing a fake ringing tone instead of music does not suffice, I really want to not answer it. The problem: it´s very hard to convince a client that a caller should go all the way to queue, thankyou, position in line, music and being picked up, even when the queue is empty. They want the caller to just ring the queue member´s phone, directly, and only go to queue() if all of them is busy. What I made so far is: 1 - try all phones before calling queue() 2 - if one is busy try the next, but if one rings until timeout, call queue() 3 - if all are busy, call queue() It works very well, but has one major flaw: the calls that get to the queue will be distributed using the queue´s strategy (random for example), but the calls that goes directly to the extensions before being queued go in a static order (roundrobin without memory) and so they will overload the first person. It would be very nice if the queue itself would distribute also this calls, using the same strategy, but passing it directly without queuing/thankyou/position-in-queue/music/etc. Part of the code I did: exten = 333,7,Dial(SIP/3000,10,) exten = 333,8,Goto(11) exten = 333,9,Dial(SIP/3027,10,) exten = 333,10,Goto(11) exten = 333,11,Dial(SIP/3001,10,) exten = 333,12,Goto(13) ; atende a fila normalmente exten = 333,13,Wait(2) exten = 333,14,Answer() exten = 333,15,Playback(queue-welcome) exten = 333,16,Queue(333|t|||0) exten = 333,108,Goto(9) exten = 333,110,Goto(11) exten = 333,112,Goto(13) Thank you, andre -- Andre Ruiz [EMAIL PROTECTED] Curitiba, PR, Brasil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Logout from queue
Via dialplan maybe? exten = xxx,1,Dial(SIP/101_Queue,20,tr) exten =xxx,2,RemoveQueueMember(Comercial_Queue,SIP/101_Queue,1) Kerry Garrison escribió: Yes, that is the functionality I am looking for, just not sure how exactly to pull that off. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Tuesday, April 25, 2006 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Auto Logout from queue Use the local channel to call the agent first, and if there is no answer, log them out. _ From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Tue 4/25/2006 2:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Auto Logout from queue i have a client that wants a function that will automatically logout an agent from a queue if they do not answer a call. This would prevent future calls from being sent to that phone if the agent forgot to logout. Any ideas? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.techdatapros.com/ http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sphinx
Ok, anyone used Sphinx with Asterisk? The docs are great at telling me how the internals of the damn thing work, but now how to USE it. I can't find a single example of how to run 'decode' in command line mode, without specifying a billion options! Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Logout from queue
that is a nice function I use a cronjob to logout everyone each evening if anyone wants that script I would love to provide it. On 4/25/06, Alberto Sagredo [EMAIL PROTECTED] wrote: Via dialplan maybe?exten = xxx,1,Dial(SIP/101_Queue,20,tr)exten =xxx,2,RemoveQueueMember(Comercial_Queue,SIP/101_Queue,1) Kerry Garrison escribió: Yes, that is the functionality I am looking for, just not sure how exactly to pull that off. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Alexander Lopez Sent: Tuesday, April 25, 2006 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Auto Logout from queue Use the local channel to call the agent first, and if there is no answer, log them out. _ From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Tue 4/25/2006 2:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Auto Logout from queue i have a client that wants a function that will automatically logout an agent from a queue if they do not answer a call. This would prevent future calls from being sent to that phone if the agent forgot to logout. Any ideas? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 -mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]http://www.techdatapros.com/ http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- $15.95/Month DreamHost Hosting SALE60 GB Disk Storage, 1.6 TB TransferTransfer Increases 16 GB Weekly http://www.dreamhost.com/r.cgi?ap0ught/shared/Use discount code caralena for $40 off all these low prices ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Touch tone recognition issues
Im experiencing touch tone recognition issues when calling some outside phone systems. For instance, if I call my Nextel phone, and try to press * to enter my voicemail, Nextels system does not hear the DTMF tone. Ive also experienced other outside phone systems for which I am unable to use their touch tone menus. Oddly, this isnt the case with all outside systems. If I call Dell, everything works great. Is this a known issue with asterisk? Im hope there is a simple setting Ive over looked. All help is appreciated. Thank you. Bryan Mahin Rediscover Personal Servicewith UNETA Please visit us @ www.uneta.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip t38 gateway tests
These people usualy hang out in the downtown area every morning waiting for work. Do you plan on actualy frying them and then transmit them over your fax machines using FoIP? You should check with your local government if it's legal. If you use just their photos then you might need written permission from them. On 4/25/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello, I patched asterisk patched with the latest t38 support . I would need some people for tests. Regards harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pressing ## end the call and return to menu
I am asterisk newbie, so please bear with me if this is an easy one. I am trying to enhance a basic calling card application to support the feature where the caller can press ## to end the current call and return to the main menu to place a new call. Any hints as to how to go about doing that? I believe part of the answer is that I would need to use the H option in the dial command, (although here the caller must dial *, and not ## to trigger the hang up). __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 56K Dialup and VOIP over same PRIs
Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. Ian -- Ian White Victoria Free-Net Association email: [EMAIL PROTECTED] http://victoria.tc.ca/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Way Audio....in the middle of a call
I experienced this today. Doing a 'show channels' in Asterisk showed a Zap line perpetually ringing the sip phone even though the sip phone was reset a few times. Doing a 'soft hangup' on the stuck Zap and the Sip allowed 2-way audio to resume. Phil Frederic Jean wrote: Hi Geoff, You might want to try tcdump, specifying the source and destination IP (to minimize the info) and see where are the RTP packets going ; you will see if they change port or something like that after a while. Cheers, Frederic - Original Message - *From:* Geoff Manning mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Tuesday, April 25, 2006 17:37 *Subject:* [Asterisk-Users] One Way Audioin the middle of a call We had a user report that they were on a SIP --- PSTN call for about 4.5 minutes before the call went to on-way audio. The user called the person back and they reported being able to hear my user, but my user couldn't hear them. The audio condition persisted for about 15 seconds before the user hung up. Where do I start to troubleshoot one way audio that occurs during a call? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 56K Dialup and VOIP over same PRIs
Ian White wrote: Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. No problem except for the detail of how to segregate the calls. There are two obvious approaches: put switching gear in front of your PM3 and Asterisk box to send calls to the correct place or have Asterisk do the switching. The mechanism we used for a similar set-up was to use a WCT411P Quad-Card set up in the following manner: Port 1: echo can enabled, slave clock, connected to PRI 1 Port 2: echo can enabled, slave clock, connected to PRI 2 Port 3: echo can disabled, master clock, cross connect to modem equipment Port 4: echo can disabled, master clock, cross connect to modem equipment Our dialplan then looks at calls coming in on the PRIs and if it was a voice DID that was dialled, handles the call directly. If it was a modem DID that was dialled, Asterisk passes the call to the modem equipment. The nice thing about the above approach (as we are told by Digium) is that the TE411P card does timeslot switching on the card so that the actual traffic in on the PRI and out to the modem equipment has no latency and, much more importantly, no opportunity for clock slips which can wreak havoc with modem calls. And the WCT411P is a lot less expensive that a six-port Adtran to do the switching in front of the Asterisk and modem equipment. Contact me directly (I'm just across the Strait from you) if you need more assistance with the above. g. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent
I've been asking about this problem in Asterisk channel... I didn't report it has a bug...Probably it is recommended... On 4/24/06, Thomas Winter [EMAIL PROTECTED] wrote:Am Wednesday 19 April 2006 16:37 schrieb Marco Mouta: How do I report a Bug to Digium? or asterisk project?Did you report this bug?I checked and have seen only an timeout in the channel will kill the deadchannels.Iam using GROUP_COUNT, so it is easy to kill my Asterisk if somebody is make some calls and disconnect the SIP-client every time.best regardsThomas___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 56K Dialup and VOIP over same PRIs
A Lucent MAX TNT will do it, there are some limitations on the TNTs ability to received caller ID name from the telco if is not sent as part of the ISDN SETUP message, many Telco's send CNAM in the FACILITY IE and the lucent ignores it. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ian White Sent: Tuesday, April 25, 2006 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 56K Dialup and VOIP over same PRIs Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. Ian -- Ian White Victoria Free-Net Association email: [EMAIL PROTECTED] http://victoria.tc.ca/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 56K Dialup and VOIP over same PRIs
Ian White wrote: Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. There are at least two options: 1) Terminate the PRIs on your Asterisk server and then drop off 'new' PRIs from the Asterisk server to the PM3; the dialplan can decide which calls to send which direction, and the T1 cards will bridge the modem calls across. 2) Terminate the PRIs on an Adtran Atlas and then have it drop off 'new' PRIs to your Asterisk server and the PM3, and it can route the incoming calls. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Way Audio....in the middle of a call
Is there an easy fix?- Dan L.On 4/25/06, Philip Edelbrock [EMAIL PROTECTED] wrote: I experienced this today.Doing a 'show channels' in Asterisk showed aZap line perpetually ringing the sip phone even though the sip phone wasreset a few times.Doing a 'soft hangup' on the stuck Zap and the Sip allowed 2-way audio to resume.PhilFrederic Jean wrote: Hi Geoff, You might want to try tcdump, specifying the source and destination IP (to minimize the info) and see where are the RTP packets going ; you will see if they change port or something like that after a while. Cheers, Frederic - Original Message - *From:* Geoff Manning mailto: [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Tuesday, April 25, 2006 17:37 *Subject:* [Asterisk-Users] One Way Audioin the middle of a call We had a user report that they were on a SIP --- PSTN call for about 4.5 minutes before the call went to on-way audio. The user called the person back and they reported being able to hear my user, but my user couldn't hear them. The audio condition persisted for about 15 seconds before the user hung up. Where do I start to troubleshoot one way audio that occurs during a call? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FastAGI Connection Failure and Hangup
Steve, you need the FastAGI contingency patch, part of the Asterisk Queues Tutorial available at http://www.orderlyq.com/asteriskqueues.html It's near the bottom of the page. Anybody know why this still hasn't made it into trunk? Matt. Steve wrote: Does anyone know how to make fastagi continue to the next priority if it can not connect to the remote AGI Server? Right now I am just getting Hangup and can't find anything on the net about this. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agents -- Extensions
How can I do the following 2 things in my dialplan? 1. find out what extension a agent is assigned to by agent id. 2. find out what agent is assigned to a extension by extension id. Anybody know how to do this? I read some where that I might have to pull it from the db. Example code is a plus :) -- ~Shaun ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Touch tone recognition issues
Bryan Mahin wrote: I’m experiencing touch tone recognition issues when calling some outside phone systems. For instance, if I call my Nextel phone, and try to press * to enter my voicemail, Nextel’s system does not “hear” the DTMF tone. I’ve also experienced other outside phone systems for which I am unable to use their touch tone menus. Oddly, this isn’t the case with all outside systems. If I call Dell, everything works great. Is this a known issue with asterisk? I’m hope there is a simple setting I’ve over looked. All help is appreciated. Thank you. Bryan Mahin Sounds like a common problem solved 20 years ago in the telephone industry, when tones were generated by a common sender and the engineer didn't read the standards and made tone duration too short. Asterisk is probably right on that edge and the tone duration needs to be somewhat longer. 75mS is probably as short as it should be, better 100 mS Of course, you didn't say much about your call routing and if Asterisk or something else is really generating the tones, so that is just a guess John Novack * * */Rediscover Personal Service with UNETA/* */Please visit us @ www.uneta.com/* ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] USB conference phone
Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 But looking for real world feedback. Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Here I am facing problem of Voice Breakage
Here I have attched file of extensions.conf please if any one have the soluition to face the problem of voice brakage. My internet E1 data line connectivity is okbecause when i wont use asterisk server then voice is clearThank YouShobhit Nirala+919871476403 SHOBHIT NIRALA CONT NO. 9871476403 How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates. extensions.conf Description: 3949034846-extensions.conf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hi...Please help me
It's all possible. Paul Hales -- Paul Hales Technical Manager Asterisk IT bus: 03 8320 8100 mob: 0434 225 491 Crazy Boy wrote: Hi Friends, I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to all of my systems. The structure for the same is: PSTN (Phone call) --- VOIP phone --- Server system --- --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) ----- ----- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using Asterisk software? If It can be implemented using Asterisk software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? Please reply me. Thank you Thanks Regards, Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://beta.messenger.yahoo.com http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://beta.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://beta.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trying to set up automatic announcement upon transfer for IVR in AAH 2.8
I am running AAH 2.8. I have an IVR for our main phone number that allows users to dial an extension directly. I would like to have a this call may be recorded announcement played before the call gets transferred. There is not a built-in option for this in the IVR web interface, but one way I can do this is to create ring groups for each user with announcements and modify the dialplan to dial the ring groups instead of the extensions. The question is, where do I do this? What part of the dialplan should I modify to make it substitute a ring group for the dialed-in extension? Sorry to post on the asterisk users list, I know AAH is not exactly related, but there is something wrong on their forum right now. I can't post there, even though I'm logged into sourceforge. Thanks, Carl ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)
John Novack wrote: Mike Garey wrote: well, the problem isn't that the card doesn't detect a disconnect, it's that it doesn't detect it immediately (or at least within a short period). Odds are that is the telco, and not the Sangoma or Digium card. That is quite normal for a 10-30 second delay. Not all telco CO's send an immediate pulse when the caller hangs up. Is there no way to detect 5-6 seconds of silence by Asterisk? This is from /path/src/asterisk/configs/voicemail.conf.sample. Amazing how much good stuff is in that directory. Especially handy to read after a significant upgrade (i.e. 1.0.x to 1.2.x) ; How many seconds of silence before we end the recording maxsilence=10 ; Silence threshold (what we consider silence, the lower, the more sensitive) silencethreshold=128 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)
Are you in the USA or Canada? Mike Garey wrote: yes, I'm using kewlstart On 4/24/06, Sean Cook [EMAIL PROTECTED] wrote: On Mon, 2006-04-24 at 17:20 -0400, Mike Garey wrote: As far as I can tell, after discussing this matter with other asterisk users in my area, my telco _does_ provide disconnect supervision.. It seems that the problem is actually related to the Sangoma A200 card I'm using, as two other people both using this same card have expressed the same problem.. Are there any other users on this list using the Sangoma A200 FXO port card, and experiencing problems with asterisk not detecting when a channel has been disconnected? Thanks, Are you using kewlstart? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] test numbers in different countries!
Hi, due to the fact that different providers need a different way to dail, I made some mistakes, whenever I changed the code. Multiple gateways, different dialing patterns, I would need some testing numbers in different countries. Testing numbers where a tape is or where a long company announcement is. Do you know such numbers? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MFCR2 in Brazil, someone?
Which version of unicall and spandsp are you using? How is your zaptel.conf and unicall.conf? []'s MM -Original Message- From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Tue, 25 Apr 2006 12:45:41 -0500 Delivered: Tue, 25 Apr 2006 11:48:34 Subject:[Asterisk-Users] MFCR2 in Brazil, someone? Does anybody have a working Asterisk server with Unicall using MFCR2 in Brazil? Were having problems. It seems SPANDSP never detect the tones from the telco. Im using brazil protocol variant. Im having lots of problems to find out why spandsp seems to not detect the MF tones. We send the first digit, the telco says they receive it, and respond with the proper signal to ask for the next digit, we just never detect the tone and the T1 timer times up. Some custom logs i have put in mfcr2.c point to spandsp r2_mf_rx always returning a zero value, what seems to mean OFF TONE, because it automatically sends the code to mf_tone_off_event() but without expecting tone because it never enters to mf_tone_on_event() something like this: OUR PBX = seize TELCO = seize ACK === == First DNIS tone == here we never detect the tone from the telco the server is Linux switch-cwb.jeffnetworks.com 2.6.9-34.ELsmp #1 SMP Thu Mar 9 06:23:23 GMT 2006 x86_64 x86_64 x86_64 GNU/Linux already tried different spandsp versions without success. Thanks in advance. -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1145987314.216908.1433.arrino.terra.com.br,5013,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] test numbers in different countries!
How about using time announments? I list of these for each country would be great! Jason Ronald Wiplinger wrote: Hi, --snip-- I would need some testing numbers in different countries. Testing numbers where a tape is or where a long company announcement is. Do you know such numbers? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hi...Please help me
On 4/24/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi Friends, [..snip..] --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) ----- ----- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using Asterisk software? If It can be implemented using Asterisk software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? [..snip..] It can be done with Asterisk. For the server side, you would need to install Asterisk on your Fedora 5 box, Zaptel and lots of Wiki reading. I don't recommend using softphones for your employee PCs. It looks like an attractive solution at first (from a cost perspective) but in reality it's not very practical (at least that was my experience). Buying 5 x 2 port ATAs will cost you around $300-$350 which is not really expensive considering the kind of powerful PBX you will have at your disposal. I would have suggested some Digium hardware for the FXS (extensions) but I think it will be a lot more expensive (for 10 extensions) than the ATAs solution. You could also look into a channel bank, but again it will be more expensive than the 5 ATAs. As for the FXO (incoming/outgoing PSTN) I recommend buying Digium hardware (TDM400P). Hope this helps, and good luck! Regards, Gonzalo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
