Re: [asterisk-users] problem starting asterisk, unable to load chan_zap
Hi, Two things come to mind, (1) being that you don't have the TE110P card jumped for an E1. (2) UDEV isn't creating the devices fast enough for the driver load. My guess is it's UDEV. You can test this theory by creating a startup script that loads the modules, put a sleep statement in that script that waits for a second or so. I have the asterisks running as a cluster managed via linux-ha. First the driver loads when booting, then I log in, and start linux-ha, which then starts asterisk. I think that is time enough for udev to create the devices, but it is not doing it in the first place. I added a rmmod wcte11xp modprobe wcte11xp to /etc/init.d/zaptel into the start section, and when I then login I have no problem to start asterisk. Sebastian ___ --Bandwidth 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] Changing the From field in Asterisk email/voicemail
Should be able to edit the following lines in /etc/asterisk/voicemail.conf ; Who the e-mail notification should appear to come from serveremail=asterisk ;[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Suber Sent: Wednesday, 6 June 2007 2:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Changing the From field in Asterisk email/voicemail Anyone know how to change the From field in Asterisk PBX voicemail/email to some other info of my choosing? ___ --Bandwidth 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] problem starting asterisk, unable to load chan_zap
Hi, Easy to check if the problem is udev: ls -l /sys/class/zaptel there are lots of subdirectories in there, at least after a rmmod modprobe. I can reboot later to see how it looks like before that. If there are files there and not under /dev/zap, udev is to blame. in /dev/zap there are a lot of devices, but as above, at least after a rmmod modprobe. I can reboot later to see how it looks like before that. What kernel version is it? What distribution? openSUSE 10.2, kernel 2.6.18.2-34-default Anyway, does Asterisk still fail? not when I do a rmmod modprobe the wcte11xp or wcte12xp respectively, without that, it fails. As which user do you run Asterisk? /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c ls -la /dev/zap shows a lot of devices: ... crw-rw 1 asterisk asterisk 196, 8 2007-06-14 07:25 8 crw-rw 1 asterisk asterisk 196, 9 2007-06-14 07:25 9 crw-rw 1 asterisk asterisk 196, 254 2007-06-14 07:25 channel crw-rw 1 asterisk asterisk 196, 0 2007-06-14 07:25 ctl crw-rw 1 asterisk asterisk 196, 255 2007-06-14 07:25 pseudo crw-rw 1 asterisk asterisk 196, 253 2007-06-14 07:25 timer kind regards Sebastian ___ --Bandwidth 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] problem starting asterisk, unable to load chan_zap
On Thu, Jun 14, 2007 at 08:10:33AM +0200, Sebastian Reitenbach wrote: Hi, Two things come to mind, (1) being that you don't have the TE110P card jumped for an E1. (2) UDEV isn't creating the devices fast enough for the driver load. My guess is it's UDEV. You can test this theory by creating a startup script that loads the modules, put a sleep statement in that script that waits for a second or so. I have the asterisks running as a cluster managed via linux-ha. First the driver loads when booting, then I log in, and start linux-ha, which then starts asterisk. I think that is time enough for udev to create the devices, but it is not doing it in the first place. I added a rmmod wcte11xp modprobe wcte11xp to /etc/init.d/zaptel into the start section, and when I then login I have no problem to start asterisk. This seems like unneeded voodoo. You have already demonstrated that ztcfg ran perfectly well after system boot with no startup. Conisder the following startup sequence: TMOUT=10 # timeout for udev to populate /dev/zap # maybe it does nothing, as the module is already loaded by # hotplug/udev ? modprobe wcte11xp # wait for /dev/zap/ctl: while [ ! -c /dev/zap/ctl ] ; do sleep 1 TMOUT=`expr $TMOUT - 1` if [ $TMOUT -eq 0 ] ; then echo Error: missing /dev/zap/ctl. udev problem? exit 1 fi done /sbin/ztcfg Basically the same as in in the zaptel init.d script but with lots of junk removed. Also, be sure to remove the lines in the modprobe configuration that run ztcfg on module startup. Otherwise you defeat the purpose of the above loop. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth 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 testing - thanx!
On 6/14/07, George Williams [EMAIL PROTECTED] wrote: Thank you both for your expert responses to my question about asterisk testing on the asterisk newsgroup. I'm taking my follow up questions off-line... 1) SIPP looks like just what I need for SIP testing, thanx. You also mentioned dialplan testing using AGI and AEL. Are these toolkits? AGI is Asterisk Gateway Interface, AEL is Asterisk Expression Language. More or less they are the same - scripts for Asterisk. 2) Using asterisk to test asterisk seems very clever. Is it possible to automate the asterisk which is acting as the tester? The reason is because I also have need for a heart-beat app - which validates the production asterisk server at regular intervals. For that you can go with simple nagios script, but if you really need to test is speech getting trough, you can use tester asterisk to connect to production (with two users), and then make call from 1st tester user to 2nd tester user. Then on 1st just play some DTMF, and listen for it with Read(). As for automation - simple cron job can place call files for asterisk. BTW, I use asterisk for several things - as a SIP proxy, the IVR, and multi-conference. So automating testing and validation of all these features is key. Regards, Atis ___ --Bandwidth 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] Queue problem
Set your core debug level to greater than 2 SET DEBUG seems not to have any effect on my asterisk. Let us know what you find. The effect was caused by an misconfigured phone: The phone did nod signal busy but ringing due to an call waiting indication. Switching off call wating indication should remove the problem. Elmar ___ --Bandwidth 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 and te110p configuration.
Dear users. My current setup uses a euroISDN E1 with 8 cahnnels for incoming and outgoing calls from a digium te110p. Currently all phones use SIP. However, I need to add some faxes lines and some POS credit card machines. These will require POTS lines with a fixed DDI. I have purchased the tdm400p and 4 FXS modules. My problem is with the zaptel.conf and zapata.conf. I am a little confused as how to separate the specific requirements for each card. How do I create a span for the tdm400p? I would imagine they require their own context and specific group? Also the channel numbers become a bit of a problem. Do they become sequential carrying on from each card I would imagine I need to modprobe the correct drivers for this card as well. Will there be any conflict? Here is my current zaptel and zapata confs with the te110p requirements ONLY. zaptel: loadzone = uk defaultzone = uk span = 1,0,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 zapata: [channels] language=en usecallerid=yes hidecallerid=no callwaiting=no callwaitingcallerid=yes restrictcid=no usecallingpres=no threewaycalling=yes callreturn=yes transfer=yes cancallforward=yes echocancelwhenbridged=yes echocancel=yes musiconhold=default rxgain=0.0 txgain=0.0 signalling=pri_cpe switchtype=euroisdn immediate=no overlapdial=yes pridialplan=unknown prilocaldialplan=unknown group=1 context = from-pstn callerid=asreceived channel = 1-8 Please would someone start me off in the right direction for adding these additional FXS devices.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the state of Asterisk Secure Remote Communications?
On 13 Jun 2007, at 22:48, Alvin Austin wrote: Hello all, The wiki has a fairly detailed description of the the issues involved with encryption of Asterisk calls: http://www.voip-info.org/wiki/view/Asterisk+encryption I'm interested in hearing what is working for people today. I think the ideal solution would be a hard phone that could be plugged in almost anywhere (dsl/cable modem, hotel, etc) and connect securely to a remote Asterisk server (both for signalling and the RTP media stream). This might be a standalone phone, or maybe one plugged into a small (broadband router sized) box. An example commercial phone system with this capability is the Mitel 3300 or SX-200 with 5xxx IP phones having teleworker capability. What solutions just work out there? (Just work means that the end user only has to know enough to plug stuff in to get a dial- tone and incoming calls). All ideas (commercial or otherwise) welcome. Well, if you aren't wedded to SIP, IAX's encryption seems to fit the bill. It 'works for me'. The design goal was to have 'good enough' encryption that didn't require any copying of keys. If you have 2 modern asterisks that are talking to each other over IAX at the moment, you can just say 'encryption=yes' in your iax.conf, and your calls between them will be encrypted. There are a few (small) provisos: 1) you have to be using password auth for the calls. 2) I don't think it works with IAX trunking 3) the called/calling numbers are in the clear. (encryption only kicks in with the 'accept' message). I don't know of any hard phones that implement it, but I guess that quite a lot of the small-asterisk-in-a-box solutions will run it just fine. I've got a pre-production version of our softphone that supports it, and I guess that iax-client probably will too soon. Tim. Thanks, Alvin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Qualify renders all SIP peers unreachable
I totally puzzled by this situation. I have asterisk 1.4.4 behind NAT. All SIP peers are working properly to place or receive calls. Any SIP peer or friend whether NATted or not will become UNREACHABLE if qualify=yes. I have identical peers on the other asterisk 1.2.16 production server. In fact, two of the phones (linksys 941 and Polycom ip500) are using one line for each asterisk. The 1.2 one works normally, the 1.4 does not. The sip confgs from sip show settings are identical on the two servers. The sip.conf peer entries were moved over exactly. Ports 5060 to 5065 are forwarded to the asterisk server. Looking at sip debug, I notice a few differences: REGISTER from phone: Authorization: Digest username=Poly, realm=asterisk,... does not show on the 1.4 server. Trying (sent by *): Supported: replaces The Via lines are the same (internal ip addresses) on both servers, but there is a Sending to 192.168... on the 1.2 message where there is none on the 1.4. What is supported: replaces ? What config setting generates the Authorization: Digest... message ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p and te110p configuration.
On Thu, Jun 14, 2007 at 09:45:01AM +0100, Matt Scott wrote: Dear users. My current setup uses a euroISDN E1 with 8 cahnnels for incoming and outgoing calls from a digium te110p. Currently all phones use SIP. However, I need to add some faxes lines and some POS credit card machines. These will require POTS lines with a fixed DDI. I have purchased the tdm400p and 4 FXS modules. *FXO* modules, right? My problem is with the zaptel.conf and zapata.conf. I am a little confused as how to separate the specific requirements for each card. How do I create a span for the tdm400p? You don't . Just 'fxsks' lines which look like bchan/dchan lines in zaptel.conf. In zapata.conf they are the same channels (with fxs_ks signalling). I would imagine they require their own context and specific group? Right. Also the channel numbers become a bit of a problem. Do they become sequential carrying on from each card cat /proc/zaptel/* I would imagine I need to modprobe the correct drivers for this card as well. Will there be any conflict? What type of conflict? The number of a channel is set when you load a driver (technically: when you register its spans to zaptel). If you want to make sure that the current channels of the E1 card keep their numbers, you should load the analog card second. On Debian systems you can guarantee that by e.g. putting the module names in the proper order in /etc/modules . Here is my current zaptel and zapata confs with the te110p requirements ONLY. zaptel: loadzone = uk defaultzone = uk span = 1,0,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 # Something of the sort of: fxsks = 32-35 zapata: [channels] language=en usecallerid=yes hidecallerid=no callwaiting=no callwaitingcallerid=yes restrictcid=no usecallingpres=no threewaycalling=yes callreturn=yes transfer=yes cancallforward=yes echocancelwhenbridged=yes echocancel=yes musiconhold=default rxgain=0.0 txgain=0.0 signalling=pri_cpe switchtype=euroisdn immediate=no overlapdial=yes pridialplan=unknown prilocaldialplan=unknown group=1 context = from-pstn callerid=asreceived channel = 1-8 ; something of the sort of: ; context = from-pots ; group = 2 cidsignalling = v23 cidstart = polarity channel = 32-35 Alternatively use genzaptelconf, but be sure to set lc_country to uk. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Addons
Hi If you use debian install the libmysqlclient-dev package David a écrit : Hello Asterisk-Users, I'm trying to install addons 1.2.6 on Asterisk 1.2.16 (is that OK?), but my MySQL server is installed on a different sever, so the MAKE of the addons fails with the following (truncated) error message: app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory. Is there any way to bypass/ignore the fact that MySQL is installed separately and enable the installation of the addons? Thanks, David Get the Yahoo! toolbar and be alerted to new email http://us.rd.yahoo.com/evt=48225/*http://new.toolbar.yahoo.com/toolbar/features/mail/index.phpwherever you're surfing. ___ --Bandwidth 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] WAV file best sound quality
thanks for you response. Am using sip to access the sound files.The sound files are recorded with higher sampling rate and 'soxed'to 8khz on the IVR machine... could it be that resampling is responsible for the degradation?Date: Wed, 13 Jun 2007 16:44:10 -0400From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [asterisk-users] WAV file best sound quality8khz is not the best sampling rate, but that is the best you can do on the PSTN. HOWEVER, you should be able to get fairly decent sound qualify out of an 8khz sound file on the phone line. We have our IVR recorded at 8khz and it sounds fine. Are you using any compression, or G711u/PSTN for your trunks/phones? On 6/13/07, Akpome Akpoguma [EMAIL PROTECTED] wrote: Hi,I have been using wav files with sample rate of 8khz and 8 bits and I find the sound quality really poor.What is the best sound quality I can achieve on Asterisk?Responses would be appreciated. Rgds,AkpomeChange is good. See what's different about Windows Live Hotmail. Check it out! ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Make every IM count. Download Windows Live Messenger and join the i’m Initiative now. It’s free. http://im.live.com/messenger/im/home/?source=TAGWL_June07___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p and te110p configuration.
I purchased FXS modules so that I could terminate the machines or faxes (eg just like a standard phone) the outgoing/incoming channel will be be provided by my E1. I hope I have the right modules for the job? - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, June 14, 2007 11:22 AM Subject: Re: [asterisk-users] TDM400p and te110p configuration. On Thu, Jun 14, 2007 at 09:45:01AM +0100, Matt Scott wrote: Dear users. My current setup uses a euroISDN E1 with 8 cahnnels for incoming and outgoing calls from a digium te110p. Currently all phones use SIP. However, I need to add some faxes lines and some POS credit card machines. These will require POTS lines with a fixed DDI. I have purchased the tdm400p and 4 FXS modules. *FXO* modules, right? My problem is with the zaptel.conf and zapata.conf. I am a little confused as how to separate the specific requirements for each card. How do I create a span for the tdm400p? You don't . Just 'fxsks' lines which look like bchan/dchan lines in zaptel.conf. In zapata.conf they are the same channels (with fxs_ks signalling). I would imagine they require their own context and specific group? Right. Also the channel numbers become a bit of a problem. Do they become sequential carrying on from each card cat /proc/zaptel/* I would imagine I need to modprobe the correct drivers for this card as well. Will there be any conflict? What type of conflict? The number of a channel is set when you load a driver (technically: when you register its spans to zaptel). If you want to make sure that the current channels of the E1 card keep their numbers, you should load the analog card second. On Debian systems you can guarantee that by e.g. putting the module names in the proper order in /etc/modules . Here is my current zaptel and zapata confs with the te110p requirements ONLY. zaptel: loadzone = uk defaultzone = uk span = 1,0,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 # Something of the sort of: fxsks = 32-35 zapata: [channels] language=en usecallerid=yes hidecallerid=no callwaiting=no callwaitingcallerid=yes restrictcid=no usecallingpres=no threewaycalling=yes callreturn=yes transfer=yes cancallforward=yes echocancelwhenbridged=yes echocancel=yes musiconhold=default rxgain=0.0 txgain=0.0 signalling=pri_cpe switchtype=euroisdn immediate=no overlapdial=yes pridialplan=unknown prilocaldialplan=unknown group=1 context = from-pstn callerid=asreceived channel = 1-8 ; something of the sort of: ; context = from-pots ; group = 2 cidsignalling = v23 cidstart = polarity channel = 32-35 Alternatively use genzaptelconf, but be sure to set lc_country to uk. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth 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] (no subject)
Hi Guy,. you should at least put a subject any way follow this link http://nerdvittles.com/index.php?p=134 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Mon, 11 Jun 2007 18:36:54 +0530 Subject: [asterisk-users] (no subject) Hi, please help me in developing and reading Text through IVR application using asterisk. can any one help me at highlevel on this, other than using SPANDSP application. Regards K.Rajesh. _ Tried the new MSN Messenger? It’s cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ With Windows Live Hotmail, you can personalize your inbox with your favorite color. www.windowslive-hotmail.com/learnmore/personalize.html?locale=en-usocid=TXT_TAGLM_HMWL_reten_addcolor_0607___ --Bandwidth 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] Sugar Auto-Dial with Asterisk?
Before I go and start coding is anyone aware of an auto-dialer plugin for Sugar CRM that will allow me to click a button when I'm in someone's account and have my phone ring and then connect me to them? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p and te110p configuration.
On 6/14/07, Matt Scott [EMAIL PROTECTED] wrote: I purchased FXS modules so that I could terminate the machines or faxes (eg just like a standard phone) the outgoing/incoming channel will be be provided by my E1. I hope I have the right modules for the job? You do indeed have the right modules for the job. FXS modules terminate phones and the like. Below is what you'll need for your basic configuration. zaptel: loadzone = uk defaultzone = uk span = 1,0,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 # note that fxs modules use fxo signaling because they're acting like the telco fxoks = 32-35 zapata: [channels] language=en usecallerid=yes hidecallerid=no callwaiting=no callwaitingcallerid=yes restrictcid=no usecallingpres=no threewaycalling=yes callreturn=yes transfer=yes cancallforward=yes echocancelwhenbridged=yes echocancel=yes musiconhold=default rxgain=0.0 txgain=0.0 signalling=pri_cpe switchtype=euroisdn immediate=no overlapdial=yes pridialplan=unknown prilocaldialplan=unknown group=1 context = from-pstn callerid=asreceived channel = 1-8 context = from-local ; again, fxs modules use fxo signaling signalling=fxo_ks cidsignalling = v23 cidstart = polarity channel = 32-35 Another thing you need to check is that the card itself is in E1 mode and wcte11xp is loaded before wctdm. ___ --Bandwidth 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] Sugar Auto-Dial with Asterisk?
Exist a module VoiceRD to do that. JuntaDeAndalucia_es_sf_diphone 2007/6/14, Matt [EMAIL PROTECTED]: Before I go and start coding is anyone aware of an auto-dialer plugin for Sugar CRM that will allow me to click a button when I'm in someone's account and have my phone ring and then connect me to them? ___ --Bandwidth 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] Adtran feature codes, extensions
Greetings, We have An Adtran 616 Total Access device talking to a colocated Asterisk machine over MGCP. Calls placed to the phones connected to the Adtran go through as do outgoing calls from the phone (prefixed by 9), but feature access codes (*97 for voicemail, for example) and extension-to-extension calls don't work. As soon as the first digit is pressed, the user hears a busy signal. I confesss to not knowing much about how MGCP works, but I can't seem to find any kind of digit map in the Adtran so is Asterisk the one listening for but not acknowledging these digits? What do I have to do to make these work? Any help appreciated. Thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.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] Sugar Auto-Dial with Asterisk?
I see that module, but it does not work with the current version of Sugar. Does anyone have a solution that works with the current version of Sugar? On 6/14/07, Nuria Fernandez [EMAIL PROTECTED] wrote: Exist a module VoiceRD to do that. JuntaDeAndalucia_es_sf_diphone 2007/6/14, Matt [EMAIL PROTECTED]: Before I go and start coding is anyone aware of an auto-dialer plugin for Sugar CRM that will allow me to click a button when I'm in someone's account and have my phone ring and then connect me to them? ___ --Bandwidth 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] Sugar Auto-Dial with Asterisk?
Try vTiger -E On 6/14/07, Matt [EMAIL PROTECTED] wrote: I see that module, but it does not work with the current version of Sugar. Does anyone have a solution that works with the current version of Sugar? On 6/14/07, Nuria Fernandez [EMAIL PROTECTED] wrote: Exist a module VoiceRD to do that. JuntaDeAndalucia_es_sf_diphone 2007/6/14, Matt [EMAIL PROTECTED]: Before I go and start coding is anyone aware of an auto-dialer plugin for Sugar CRM that will allow me to click a button when I'm in someone's account and have my phone ring and then connect me to them? ___ --Bandwidth 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 -- Best regards, Ed Pimentel AgileCO Founder Mail: edpimentl[at]gmail.com Mail2: edpimentl[at]ieee.org IM: edpimentl [AOL | Jabber | Yahoo | MSN ] Voip: edpimentl [SKype | GoogleTalk ] Mobile Content Marketing/Management/Digital Delivery http://mobilecentral.ws Mobile ( Context Aware, AmbientIntelligence, Location ) based Social Network http://TagR.mobi (Alpha) Mobile Payment - P2P Payment http://goowallet.ws http://agilepay.ws [S4]Secure Scalable Streaming Storage GridService http://DatR.ws Sponsor of P2PSIP open source [viasip_ng] project Based on IETF P2PSIP WG https://sourceforge.net/projects/viasip/ http://groups.google.com/group/viasip_ng ___ --Bandwidth 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: Slightly OT:CSU on Digium cards, and it's requirement
On 6/13/07, Erik Anderson [EMAIL PROTECTED] wrote: On 6/13/07, C F [EMAIL PROTECTED] wrote: This is just weird I wrote it in caps so you can read it but you still didn't read it so here it is again: its a T1 card that does NOT have a CSU in it, and it is working fine and yes it is a T1 providing PRI. sarcasm Dang shmaltz. You've convinced us - we've all been wasting our precious money on CSUs this whole time. We're all idiots! /sarcasm Seriously - if you're so sure about your card not having a CSU, what is the make/model? Pony up, man. It's a Panasonic KX-TA0187 for T1, or KX-TA02290 The docs and technicians say it doesn have one AND that the FCC requires it. Hence my qeustion does the FCC require it. -Erik ___ --Bandwidth 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: Slightly OT:CSU on Digium cards, and it's requirement
On 6/13/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting C F [EMAIL PROTECTED]: This is just weird I wrote it in caps so you can read it but you still didn't read it so here it is again: its a T1 card that does NOT have a CSU in it, and it is working fine and yes it is a T1 providing PRI. If you are convinced that's what you have, and it works then great. As far as the fcc requirement, who do you think is going to be on the short end of the stick when you have a line issue and the telco wants to do a loopback test and you have no csu to loop ? I highly doubt they are going to fine you or anything like that, but you'll certainly get a bill if they have to come on site because you don't have something there that allows them to test remotely - fcc requirement or not, its only fair don't you think ? I agree with you, but I am just checking to make sure if it is or not an FCC requirement. On 6/11/07, Alex Balashov [EMAIL PROTECTED] wrote: On Mon, 11 Jun 2007, C F wrote: I disagree with this, I have several T1s that don't use Digium equipment and are directly connecting to T1 cards that DONT have a CSU and work fine. The reason this thing came up was because I was going thru documentation for such a card and it mentioned it's an FCC requirement. That's not possible, unless the handoff you're getting is not actually T1. However, the card almost certainly has a very seamless, inline/onboard CSU of which you aren't even aware of. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth 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: Slightly OT:CSU on Digium cards, and it's requirement
On 6/13/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Erik Anderson [EMAIL PROTECTED]: On 6/13/07, C F [EMAIL PROTECTED] wrote: This is just weird I wrote it in caps so you can read it but you still didn't read it so here it is again: its a T1 card that does NOT have a CSU in it, and it is working fine and yes it is a T1 providing PRI. sarcasm Dang shmaltz. You've convinced us - we've all been wasting our precious money on CSUs this whole time. We're all idiots! /sarcasm Seriously - if you're so sure about your card not having a CSU, what is the make/model? Pony up, man. I'll bet that alarm light on it is for something else right ? Yes it's there for when it's not synched, but no diags. -Erik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth 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] Linksys SPA941
Dear Group, I have just purchased two Linksys SPA941 and flashed these to the latest firmware. Everything works well except for the Hold button? Has anyone else experienced the same issue? What was the solution? Kind Regards Shad Mortazavi ___ --Bandwidth 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] ESI Phone System Integration
ESI Phone systems are supposed to support IP stations via SIP integration(http://www.esi-estech.com/products/systems/ESICS/), has anyone ever tried to link Asterisk with one of these? I'm thinking my asterisk box could be an extension off that phone system, that would then provide a Dial by Name directory to use. Not elegant, but it'd work. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event
Lee Jenkins wrote: Hi all, My company has pretty much standardized on Polycom phones and I am in the beginning phase of writing a GUI for administering/managing polycom provisioning at multiple sites which we intend to release as OS. I've started studying the docs and I'm having trouble understanding the following xml attribute: voIpProt.SIP.requestValidation.x.request.y.event I understand what it does (at least conceptually) but ss the x variable still referring to a server (1 or 2)? And the y var, what is it referring to? An event? Which one? Determines which events specified with the Event header should be validated; only applicable when voIp- Prot.SIP.requestValidation.x.request is set to “SUBSCRIBE” or “NOTIFY”. If set to Null, all events will be validated. Please excuse me if it's an obvious question. Just as a breadcrumb, here is what Polycom support says of the this portion of the XML config: The voIpProt.SIP.requestValidation parameter is used for validation. The validation is used for security purposes. To set it up properly, you have the following parameters involved which compose the overall validation feature. The parameters involved are: voIpProt.SIP.requestValidation.x.request voIpProt.SIP.requestValidation.x.method voIpProt.SIP.requestValidation.x.request.y.event voIpProt.SIP.requestValidation.digest.realm Explanation: voIpProt.SIP.requestValidation.x.request With this parameter, you can specify which methods you want the phone to validate. The list of methods allowed as values are listed in the Admin Guide. Ex: if you wanted to use validation against all INVITES, this parameter would look like this voIpProt.SIP.requestValidation.INVITE.request voIpProt.SIP.requestValidation.x.method This parameter defines the method of validation to be used. The list of methods allowed as values are listed in the Admin Guide. The three methods are source, digest or both. If you wanted to use source as the method, the parameter would like this voIpProt.SIP.requestValidation.source.method. This means that when the phone is using voIpProt.SIP.requestValidation.INVITE.request it will apply voIpProt.SIP.requestValidation.source.method and validate that the INVITE is coming from the IP address specified on its line registration. voIpProt.SIP.requestValidation.x.request.y.event This parameter is only used when you specify voIpProt.SIP.requestValidation.x.request to be voIpProt.SIP.requestValidation.SUBSCRIBE.request or voIpProt.SIP.requestValidation.NOTIFY.request. This parameter will also do validation based on the method used on voIpProt.SIP.requestValidation.x.method against the EVENTS within a “SUBSCRIBE” or “NOTIFY”. Since the RFC for SIP may have different events, the list is not provided in the admin guide. For an updated list of EVENTS please check the RFC. A less updated list of EVENTS used within a NOTIFY is as follows: conference dialog message-summary presence refer reg winfo voIpProt.SIP.requestValidation.digest.realm In this parameter you can specify a string which you have also specified on your server. The value can be any valid string. Once the phone advertises the string, the server will match it against is list of allowed users challenging the phone for an user name and password using authentication digest. Generally, this string contains the name of the host performing the authentication. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sugar Auto-Dial with Asterisk?
Thanks for the suggestion, unfortunately we are using SugarCRM. On 6/14/07, EdPimentl [EMAIL PROTECTED] wrote: Try vTiger -E On 6/14/07, Matt [EMAIL PROTECTED] wrote: I see that module, but it does not work with the current version of Sugar. Does anyone have a solution that works with the current version of Sugar? On 6/14/07, Nuria Fernandez [EMAIL PROTECTED] wrote: Exist a module VoiceRD to do that. JuntaDeAndalucia_es_sf_diphone 2007/6/14, Matt [EMAIL PROTECTED]: Before I go and start coding is anyone aware of an auto-dialer plugin for Sugar CRM that will allow me to click a button when I'm in someone's account and have my phone ring and then connect me to them? ___ --Bandwidth 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 -- Best regards, Ed Pimentel AgileCO Founder Mail: edpimentl[at]gmail.com Mail2: edpimentl[at]ieee.org IM: edpimentl [AOL | Jabber | Yahoo | MSN ] Voip: edpimentl [SKype | GoogleTalk ] Mobile Content Marketing/Management/Digital Delivery http://mobilecentral.ws Mobile ( Context Aware, AmbientIntelligence, Location ) based Social Network http://TagR.mobi (Alpha) Mobile Payment - P2P Payment http://goowallet.ws http://agilepay.ws [S4]Secure Scalable Streaming Storage GridService http://DatR.ws Sponsor of P2PSIP open source [viasip_ng] project Based on IETF P2PSIP WG https://sourceforge.net/projects/viasip/ http://groups.google.com/group/viasip_ng ___ --Bandwidth 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: Slightly OT:CSU on Digium cards, and it's requirement
Quoting C F [EMAIL PROTECTED]: On 6/13/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting C F [EMAIL PROTECTED]: This is just weird I wrote it in caps so you can read it but you still didn't read it so here it is again: its a T1 card that does NOT have a CSU in it, and it is working fine and yes it is a T1 providing PRI. If you are convinced that's what you have, and it works then great. As far as the fcc requirement, who do you think is going to be on the short end of the stick when you have a line issue and the telco wants to do a loopback test and you have no csu to loop ? I highly doubt they are going to fine you or anything like that, but you'll certainly get a bill if they have to come on site because you don't have something there that allows them to test remotely - fcc requirement or not, its only fair don't you think ? I agree with you, but I am just checking to make sure if it is or not an FCC requirement. if nothing else its definately an implied requirement since for sure there is a requirement that the terminal equipment must be compatible with the CO equipment, and without a CSU its not going to work at all. (whether the vendor calls it a csu or not, there is still something doing that job) So if it works there is a csu of some sort in there regardless of what the marketing material says. On 6/11/07, Alex Balashov [EMAIL PROTECTED] wrote: On Mon, 11 Jun 2007, C F wrote: I disagree with this, I have several T1s that don't use Digium equipment and are directly connecting to T1 cards that DONT have a CSU and work fine. The reason this thing came up was because I was going thru documentation for such a card and it mentioned it's an FCC requirement. That's not possible, unless the handoff you're getting is not actually T1. However, the card almost certainly has a very seamless, inline/onboard CSU of which you aren't even aware of. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth 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: Slightly OT:CSU on Digium cards, and it's requirement
Quoting C F [EMAIL PROTECTED]: On 6/13/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Erik Anderson [EMAIL PROTECTED]: On 6/13/07, C F [EMAIL PROTECTED] wrote: This is just weird I wrote it in caps so you can read it but you still didn't read it so here it is again: its a T1 card that does NOT have a CSU in it, and it is working fine and yes it is a T1 providing PRI. sarcasm Dang shmaltz. You've convinced us - we've all been wasting our precious money on CSUs this whole time. We're all idiots! /sarcasm Seriously - if you're so sure about your card not having a CSU, what is the make/model? Pony up, man. I'll bet that alarm light on it is for something else right ? Yes it's there for when it's not synched, but no diags. that is a function of the csu, so you have one. -Erik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth 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: Slightly OT:CSU on Digium cards, and it's requirement
I probably shouldn't be hijacking this thread but it seems that there's some people paying attention here that know what they're talking about. We've recently acquired a cisco IAD 2400 router with 2MFT-T1 VWIC card in it. Doing some cursory reading It seems that this card can be interfaced with a PRI. (I really DON'T know what I'm talking about here so my terminology might be all wrong). My question is this: If we want to get an analog trunk into the building and interface that trunk with the 2MFT card, can we then use asterisk to receive/send calls over this cisco router? How would this be accomplished? Do the cisco routers take calls via SIP or is there some other mechanism to pass calls off through this card. The reason I ask is that my boss is a cheap bastard and wants to avoid spending the $'s on a digium card if possible. 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
Re: [asterisk-users] SIP Options Reply Ignored
I am seeing this too on both Polycom and Linksys phones, as well as external SIP peerns not behind NAT, such as FWD. I've posted a couple of times about it, but I don't see the posts. On 6/3/07, Ian Clough [EMAIL PROTECTED] wrote: Hi I have FC6 system in the office running SVN-trunk-r63567 It is behind a NAT router which I have configured to do port forwarding etc. Asterisk connects and registers correctly to my SIP service (Sipgate.co.uk) and I can make and receive calls from any SIP phone on the office LAN. The problem comes when I try to use a SIP phone at home (also behind a NAT router). The phone registers correctly and I can see the SIP OPTONS packets being sent to the phone (SNOM 190). I can see an OK reply being received by Asterisk (using SIP DEBUG). However the OK reply appears to be ignored and a retransmission is made and the phone is marked as UNREACHABLE and will not accept any calls. Any ideas? Ian C ___ --Bandwidth 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] real-time HINTS
I noticed that there is a function in the func_odbc.conf called PRESENCE exists. I am assuming that this goes into dial plan but it is not clear how this might be used. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending text to a phone that's no in-use ...
Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, Hello World) but that doesn't exist ... I'm after an application where someone (say a receptionist) can send one of a small set of pre-defined messages, so that when someone next looks at their phone it might say: Wife Called, Check Email, collect printout, and so on, rather than leaving voicemail, but I'm not sure it's actually possible... Failing that, I suppose I could just dump a pre-canned audio file the appropriate header file in the voicemail spool.. but I'd rather just send a message to the phone. Cheers, Gordon ___ --Bandwidth 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] Sending text to a phone that's no in-use ...
If they're polycom 501s or higher, you could have each phone use a different homepage. Those pages could be loaded dynamically (say in php) and then you could just store the message to display in a database. Probably easiest to maintain and create an interface for. Rob Gordon Henderson wrote: Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, Hello World) but that doesn't exist ... I'm after an application where someone (say a receptionist) can send one of a small set of pre-defined messages, so that when someone next looks at their phone it might say: Wife Called, Check Email, collect printout, and so on, rather than leaving voicemail, but I'm not sure it's actually possible... Failing that, I suppose I could just dump a pre-canned audio file the appropriate header file in the voicemail spool.. but I'd rather just send a message to the phone. Cheers, Gordon ___ --Bandwidth 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] ODBC voicemail questions
Before I head down the path of converting voicemail to an ODBC backend, I have a couple questions that I was hoping someone would know. 1. Is the voicemail message stored in the datbase, or just it's location/filename? 2. Does MWI propagate when using an ODBC backend? 3. If it does both of those things, wouldn't it work well for a centralized voicemail system instead of a solution like NFS or weird rsync scripts? -- Kyle Sexton ___ --Bandwidth 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] Sending text to a phone that's no in-use ...
On Thu, 14 Jun 2007, Rob Schall wrote: If they're polycom 501s or higher, you could have each phone use a different homepage. Those pages could be loaded dynamically (say in php) and then you could just store the message to display in a database. Probably easiest to maintain and create an interface for. Snom's or Grandstream GXP2000's I'm afraid... Sending text to them while in a call works fine (although reading the text on the Snom was challenging until I'd worked out I needed to reprogram one of the function keys :) Thanks, Gordon Rob Gordon Henderson wrote: Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, Hello World) but that doesn't exist ... I'm after an application where someone (say a receptionist) can send one of a small set of pre-defined messages, so that when someone next looks at their phone it might say: Wife Called, Check Email, collect printout, and so on, rather than leaving voicemail, but I'm not sure it's actually possible... Failing that, I suppose I could just dump a pre-canned audio file the appropriate header file in the voicemail spool.. but I'd rather just send a message to the phone. Cheers, Gordon ___ --Bandwidth 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] Linksys SPA941
We had several of these when we were first playing around with Asterisk. They are somewhat nice. The audio quality left some to be desired, however, we did not have a hold button issue. On 6/14/07, Shad Mortazavi [EMAIL PROTECTED] wrote: Dear Group, I have just purchased two Linksys SPA941 and flashed these to the latest firmware. Everything works well except for the Hold button? Has anyone else experienced the same issue? What was the solution? Kind Regards Shad Mortazavi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA941
On Thursday June 14 2007 1:12 pm, Matt wrote: We had several of these when we were first playing around with Asterisk. They are somewhat nice. The audio quality left some to be desired, however, we did not have a hold button issue. On 6/14/07, Shad Mortazavi [EMAIL PROTECTED] wrote: Dear Group, I have just purchased two Linksys SPA941 and flashed these to the latest firmware. Everything works well except for the Hold button? Has anyone else experienced the same issue? What was the solution? Kind Regards Shad Mortazavi I just installed 64 of these for a customer and the hold works on the ones that have been tested. We are not on the latest firmware yet though. I will be testing that tomorrow. John M -- John Millican Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 Phone (603) 764-9163 ___ --Bandwidth 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] Sending text to a phone that's no in-use ...
I think you mean 60x not 50x. The polycom 501s don't have the microbrowser. Rob Schall wrote: If they're polycom 501s or higher, you could have each phone use a different homepage. Those pages could be loaded dynamically (say in php) and then you could just store the message to display in a database. Probably easiest to maintain and create an interface for. Rob Gordon Henderson wrote: Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, Hello World) but that doesn't exist ... I'm after an application where someone (say a receptionist) can send one of a small set of pre-defined messages, so that when someone next looks at their phone it might say: Wife Called, Check Email, collect printout, and so on, rather than leaving voicemail, but I'm not sure it's actually possible... Failing that, I suppose I could just dump a pre-canned audio file the appropriate header file in the voicemail spool.. but I'd rather just send a message to the phone. Cheers, Gordon ___ --Bandwidth 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] WAV file best sound quality
On 6/13/07, Akpome Akpoguma [EMAIL PROTECTED] wrote: I have been using wav files with sample rate of 8khz and 8 bits and I find the sound quality really poor. 8khz is correct, if you are using 8 bits, you need to use 16 bits if I'm not mistaken. ___ --Bandwidth 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] Sending text to a phone that's no in-use ...
I believe the newer versions of firmware do implement the microbrowser on the 501. - Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: I think you mean 60x not 50x. The polycom 501s don't have the microbrowser. Rob Schall wrote: If they're polycom 501s or higher, you could have each phone use a different homepage. Those pages could be loaded dynamically (say in php) and then you could just store the message to display in a database. Probably easiest to maintain and create an interface for. Rob Gordon Henderson wrote: Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, Hello World) but that doesn't exist ... I'm after an application where someone (say a receptionist) can send one of a small set of pre-defined messages, so that when someone next looks at their phone it might say: Wife Called, Check Email, collect printout, and so on, rather than leaving voicemail, but I'm not sure it's actually possible... Failing that, I suppose I could just dump a pre-canned audio file the appropriate header file in the voicemail spool.. but I'd rather just send a message to the phone. Cheers, Gordon ___ --Bandwidth 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 -- Jason Parker Digium ___ --Bandwidth 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] Sending text to a phone that's no in-use ...
Actually, sorry to not research this first: 14759: Added microbrowser support to the SoundPoint IP 501 platform from http://www.voip-info.org/wiki/index.php?page=Polycom+Microbrowser But I'm not sure which SIP firmware this is talking about being present in. Mojo with Horan Company, LLC wrote: I think you mean 60x not 50x. The polycom 501s don't have the microbrowser. Rob Schall wrote: If they're polycom 501s or higher, you could have each phone use a different homepage. Those pages could be loaded dynamically (say in php) and then you could just store the message to display in a database. Probably easiest to maintain and create an interface for. Rob Gordon Henderson wrote: Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, Hello World) but that doesn't exist ... I'm after an application where someone (say a receptionist) can send one of a small set of pre-defined messages, so that when someone next looks at their phone it might say: Wife Called, Check Email, collect printout, and so on, rather than leaving voicemail, but I'm not sure it's actually possible... Failing that, I suppose I could just dump a pre-canned audio file the appropriate header file in the voicemail spool.. but I'd rather just send a message to the phone. Cheers, Gordon ___ --Bandwidth 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] Sending text to a phone that's no in-use ...
Actually they do, but only if you're running SIP firmware 2.1 or higher. Mojo with Horan Company, LLC wrote: I think you mean 60x not 50x. The polycom 501s don't have the microbrowser. Rob Schall wrote: If they're polycom 501s or higher, you could have each phone use a different homepage. Those pages could be loaded dynamically (say in php) and then you could just store the message to display in a database. Probably easiest to maintain and create an interface for. Rob Gordon Henderson wrote: Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, Hello World) but that doesn't exist ... I'm after an application where someone (say a receptionist) can send one of a small set of pre-defined messages, so that when someone next looks at their phone it might say: Wife Called, Check Email, collect printout, and so on, rather than leaving voicemail, but I'm not sure it's actually possible... Failing that, I suppose I could just dump a pre-canned audio file the appropriate header file in the voicemail spool.. but I'd rather just send a message to the phone. Cheers, Gordon ___ --Bandwidth 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] Sending text to a phone that's no in-use ...
Awesome, thanks for this tip! Moj Dave Fullerton wrote: Actually they do, but only if you're running SIP firmware 2.1 or higher. Mojo with Horan Company, LLC wrote: I think you mean 60x not 50x. The polycom 501s don't have the microbrowser. Rob Schall wrote: If they're polycom 501s or higher, you could have each phone use a different homepage. Those pages could be loaded dynamically (say in php) and then you could just store the message to display in a database. Probably easiest to maintain and create an interface for. Rob Gordon Henderson wrote: Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, Hello World) but that doesn't exist ... I'm after an application where someone (say a receptionist) can send one of a small set of pre-defined messages, so that when someone next looks at their phone it might say: Wife Called, Check Email, collect printout, and so on, rather than leaving voicemail, but I'm not sure it's actually possible... Failing that, I suppose I could just dump a pre-canned audio file the appropriate header file in the voicemail spool.. but I'd rather just send a message to the phone. Cheers, Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WAV file best sound quality
Ahh I didn't see that in the first post. Yes Mr. SpamSucks is correct. You should use 8khz @ 16bits. Using 8khz @ 8bits will sound like a drowning goat under water. On 6/14/07, randulo [EMAIL PROTECTED] wrote: On 6/13/07, Akpome Akpoguma [EMAIL PROTECTED] wrote: I have been using wav files with sample rate of 8khz and 8 bits and I find the sound quality really poor. 8khz is correct, if you are using 8 bits, you need to use 16 bits if I'm not mistaken. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk GUI
Hi List; Where I can download Asterisk GUI and what I can have benifit from it? Regards Bilal Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ --Bandwidth 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] Qualify renders all SIP peers unreachable
What does sip show peers output ? Also set a timeout in millisec like qualify=200 instead of qualify=yes On 14/06/07, randulo [EMAIL PROTECTED] wrote: I totally puzzled by this situation. I have asterisk 1.4.4 behind NAT. All SIP peers are working properly to place or receive calls. Any SIP peer or friend whether NATted or not will become UNREACHABLE if qualify=yes. I have identical peers on the other asterisk 1.2.16 production server. In fact, two of the phones (linksys 941 and Polycom ip500) are using one line for each asterisk. The 1.2 one works normally, the 1.4 does not. The sip confgs from sip show settings are identical on the two servers. The sip.conf peer entries were moved over exactly. Ports 5060 to 5065 are forwarded to the asterisk server. Looking at sip debug, I notice a few differences: REGISTER from phone: Authorization: Digest username=Poly, realm=asterisk,... does not show on the 1.4 server. Trying (sent by *): Supported: replaces The Via lines are the same (internal ip addresses) on both servers, but there is a Sending to 192.168... on the 1.2 message where there is none on the 1.4. What is supported: replaces ? What config setting generates the Authorization: Digest... message ? ___ --Bandwidth 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] Que on A2Billing
Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the IVR Prompt... Meaning from my XLite dialer I want to dial directly and let A2Billing do the billing part. Right now is something like when I dial any number from XLite, A2Billing script is invoked and it will announce You have XXX amount, please enter the number you wish to call followed by #. And then I have to enter the number again and then the call is initiated... Its kinda annoying to do that every time you want to call. Is there anyway to modify config some where, so it will do the billing in background when the phone call is hangup. Cheers, Nitesh ___ --Bandwidth 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: Slightly OT:CSU on Digium cards, and it's requirement
On Thu, 14 Jun 2007, C F wrote: On 6/13/07, Erik Anderson [EMAIL PROTECTED] wrote: On 6/13/07, C F [EMAIL PROTECTED] wrote: This is just weird I wrote it in caps so you can read it but you still didn't read it so here it is again: its a T1 card that does NOT have a CSU in it, and it is working fine and yes it is a T1 providing PRI. sarcasm Dang shmaltz. You've convinced us - we've all been wasting our precious money on CSUs this whole time. We're all idiots! /sarcasm Seriously - if you're so sure about your card not having a CSU, what is the make/model? Pony up, man. It's a Panasonic KX-TA0187 for T1, or KX-TA02290 The docs and technicians say it doesn have one AND that the FCC requires it. Hence my qeustion does the FCC require it. I think what he's referring to is really the KX-TD187... which is a T1 interface module for the Panasonic KX-TD1232 Digital Hybrid Phone System (I have one of these systems, but not the T1 module). Now there is a KX-TA1232 analog system, and maybe there was a KX-TA187 module for it that has since been discontinued... but I think he meant the digital one. http://www.ablecomm.com/t1isdideq.html They do SAY it doesn't have a CSU... but it's beyond my understanding of how it could possibly work without one. They seem to sell a separate CSU module that can go with it. Maybe he's just not seeing the extra little box because there's more wire between that and the demarc? Was this a system that was already installed for you? Or did you install it yourself? Maybe the CSU is external and you just didn't recognize/see it there? When I first started working with T1's, most CSU's were external. I still have several of them in storage in fact... and I still use external CSU/DSU's on my production network today. :-) I'm typing this message and it will be sent over a T1 connected to 2 external CSU's before it reaches the internet. Bottom line is, no matter what the FCC says... and if somehow you managed to get it to work without a CSU... I believe the phone company would have a fit if they knew you connected equipment to their network without a CSU on it. They're very big on standards-compliance and stuff like that. Sometime look into their rules and regs about colocating equipment inside one of their CO's... it's very very strict. -- Nick ___ --Bandwidth 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] b410p
Hello, I'm trying to set up a b410p rdsi card, and I'm having problems getting it up. I followed the instruction on asteriskguru and everything seem to be fine but all leds on the card are in red. [EMAIL PROTECTED] ~]# uname -a Linux rdsipbx 2.6.15.7 #2 Tue Jun 5 16:37:07 CEST 2007 i686 i686 i386 GNU/Linux [EMAIL PROTECTED] ~]# dmesg |grep Digium HFC-multi: card manufacturer: 'Cologne Chip AG' card name: 'HFC-4S Digium Card' clock: normal [EMAIL PROTECTED] ~]# cat /etc/misdn-init.conf card=1,0x4 te_ptp=1,2,3,4 poll=128 dsp_options=0 debug=0xf [EMAIL PROTECTED] ~]# asterisk -V Asterisk 1.2.18 *CLI misdn show stacks BEGIN STACK_LIST: * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 * Port 2 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 * Port 3 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 * Port 4 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 *CLI misdn show channels Chan List: (nil) the service provider is TELEFONICA in SPAIN. any idea? -- Luis José Da Silva G. http://luisjose.nelug.org.ve/ GPG ID: DE0CAC65 GPG Key Fingerprint: DAC3 4F56 CACF EA15 2729 F3FF C3B1 7990 DE0C AC65 ___ --Bandwidth 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 GUI
On 6/14/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; Where I can download Asterisk GUI and what I can have benifit from it? Whaddya know - there's a whole page on the wiki dedicated to such things: http://www.voip-info.org/wiki-Asterisk+GUI ;-) I'm a CLI-only guy myself, so I can't comment on the quality (or lack therof) of any of the offerrings. I believe FreePBX is the most popular by far, though, so you may want to check that out first. -erik ___ --Bandwidth 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] Qualify renders all SIP peers unreachable
On 6/14/07, Jaswinder Singh [EMAIL PROTECTED] wrote: What does sip show peers output ? Also set a timeout in millisec like qualify=200 instead of qualify=yes Doesn't matter. I've used qualify=2000 There is another thread about this now, OPTIONS response from the phone is ignored. ___ --Bandwidth 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] Sending text to a phone that's no in-use ...
Yes they do. If you download the newest boot rom, they sure do have it now. I was surprised myself when I saw the feature, but we use it on all of our 501s here. The resolution isn't pretty, but it works. :) Rob Mojo with Horan Company, LLC wrote: I think you mean 60x not 50x. The polycom 501s don't have the microbrowser. Rob Schall wrote: If they're polycom 501s or higher, you could have each phone use a different homepage. Those pages could be loaded dynamically (say in php) and then you could just store the message to display in a database. Probably easiest to maintain and create an interface for. Rob Gordon Henderson wrote: Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, Hello World) but that doesn't exist ... I'm after an application where someone (say a receptionist) can send one of a small set of pre-defined messages, so that when someone next looks at their phone it might say: Wife Called, Check Email, collect printout, and so on, rather than leaving voicemail, but I'm not sure it's actually possible... Failing that, I suppose I could just dump a pre-canned audio file the appropriate header file in the voicemail spool.. but I'd rather just send a message to the phone. Cheers, Gordon ___ --Bandwidth 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] Re: Slightly OT:CSU on Digium cards, and it's requirement
On Thu, 14 Jun 2007, Nick Seraphin wrote: Bottom line is, no matter what the FCC says... and if somehow you managed to get it to work without a CSU... I believe the phone company would have a fit if they knew you connected equipment to their network without a CSU on it. They're very big on standards-compliance and stuff like that. Sometime look into their rules and regs about colocating equipment inside one of their CO's... it's very very strict. Well, moreover, sans CSU the technology would not be T1. :-) I imagine there's probably a DSX-1 interface to an external CSU somewhere, or else another CSU card or element in the chassis. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth 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 GUI
Voip-info has some different links to packages out there for a gui based asterisk. In my experience, I've found it much easier to tweak a dialplan and user accounts by hand. We are using realtime/mysql for all our voicemail/sip/extensions, and I have a small gui I made that creates those initial entries, but afterwards, I do the alterations by hand as it gives me more control and you don't have to rely on apache going down. I also use asterisk cdr, which is a great gui if you are storing your cdr records in mysql as well. It generates some decent graphs and break downs on usage and has a decent search tool. bilal ghayyad wrote: Hi List; Where I can download Asterisk GUI and what I can have benifit from it? Regards Bilal Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ --Bandwidth 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] Que on A2Billing
On Thu, 2007-06-14 at 14:46 -0400, Nitesh Divecha wrote: Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the IVR Prompt... Meaning from my XLite dialer I want to dial directly and let A2Billing do the billing part. Right now is something like when I dial any number from XLite, A2Billing script is invoked and it will announce You have XXX amount, please enter the number you wish to call followed by #. And then I have to enter the number again and then the call is initiated... Its kinda annoying to do that every time you want to call. Is there anyway to modify config some where, so it will do the billing in background when the phone call is hangup. Yes, is possible using the a2billing.conf file in the right way. I don't have the v1.3 installed, but in the previous release 1.2.3 you must have to modify : use_dnid=YES number_try=1 say_balance_after_auth=NO say_balance_after_call=NO say_rateinitial=NO say_timetocall=NO Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth 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] Sending text to a phone that's no in-use ...
Thanks, everybody, for bringing this to my attention! I can't wait to play around with it! Moj Rob Schall wrote: Yes they do. If you download the newest boot rom, they sure do have it now. I was surprised myself when I saw the feature, but we use it on all of our 501s here. The resolution isn't pretty, but it works. :) Rob Mojo with Horan Company, LLC wrote: I think you mean 60x not 50x. The polycom 501s don't have the microbrowser. Rob Schall wrote: If they're polycom 501s or higher, you could have each phone use a different homepage. Those pages could be loaded dynamically (say in php) and then you could just store the message to display in a database. Probably easiest to maintain and create an interface for. Rob Gordon Henderson wrote: Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, Hello World) but that doesn't exist ... I'm after an application where someone (say a receptionist) can send one of a small set of pre-defined messages, so that when someone next looks at their phone it might say: Wife Called, Check Email, collect printout, and so on, rather than leaving voicemail, but I'm not sure it's actually possible... Failing that, I suppose I could just dump a pre-canned audio file the appropriate header file in the voicemail spool.. but I'd rather just send a message to the phone. Cheers, Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question on capacity
Can one server (like AMD 6000+ X2) with 2 GIG ram running asterisk 1.4 handle having 2100 wireless phones connected. All phones will not be talking at the same time only a couple will be. There may be 1 T1 card in the box. Will this work? If not how does one handle this situation. Thanks, 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
Re: [asterisk-users] question on capacity
Jerry Geis wrote: Can one server (like AMD 6000+ X2) with 2 GIG ram running asterisk 1.4 handle having 2100 wireless phones connected. All phones will not be talking at the same time only a couple will be. There may be 1 T1 card in the box. Will this work? If not how does one handle this situation. Any transcoding involved, or will you be using G711u on all the phones? How many, max, do you see talking at the same time? What will the reregistration interval be for the phones (How often will they check into the Asterisk box)? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on capacity
Jerry Geis wrote: Can one server (like AMD 6000+ X2) with 2 GIG ram running asterisk 1.4 handle having 2100 wireless phones connected. All phones will not be talking at the same time only a couple will be. There may be 1 T1 card in the box. Will this work? If not how does one handle this situation. yes, it can work, but not with asterisk alone. SIP phones consume a lot of resources in asterisk. Better is a setup with a combo of (open)ser and asterisk, basically asterisk will be a gateway to the fixed phone net... Thanks, 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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth 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/R2 line dropped immediately after connect
Hi, I am trying to set up an E1 line with CAS signaling using available unicall patches with libmfcr2 implementation. Inbound calls works well, I am able to get DNIS and ANI from incoming call, but I am still not able to make an outbound call with our local carrier. After tweaking of protocolvariant parameter in unicall.conf I was able to find proper values for inbound and outbound calls, but when outbound connection is created, it is immediately dropped by opposite side immediately after connection is made. I think, that we're receiving billing pulses on the line and they're interpreted as a cleaning request. I found similar problem in the list before: http://lists.digium.com/pipermail/asterisk-users/2006-June/156732.html. Steve suggested to use some kind of timeout to workaround this problem. I've tried to change many timeouts in libmfcr2 library, without any success. Can somebody help me to go around this line drop-down problem? Relevant part of debug log file: Jun 9 15:19:27 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 - 6 off [2/ 40/Group I /DNIS] Jun 9 15:19:27 WARNING[19398]: chan_unicall.c:2644 handle_uc_event: Unicall/21 event Alerting Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 - 0101 [1/ 200/Await answer /DNIS] Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:2644 handle_uc_event: Unicall/21 event Connected -- UniCall/21-1 answered UniCall/55-1 Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:1411 unicall_answer: Answer Call Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Call control(5) Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Answer call Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 0101 - [1/ 20/Group B /Accepted Paid] -- Attempting native bridge of UniCall/55-1 and UniCall/21-1 Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:1620 unicall_bridge: unicall_bridge called Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:2644 handle_uc_event: Unicall/55 event Answered Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 - 0001 [1/ 400/Answered /DNIS] Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 R2 prot. err. [1/ 400/Answered /DNIS] cause 32773 - Unexpected CAS bit pattern Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 1001 - [1/ 1/Idle /Idle] Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:2644 handle_uc_event: Unicall/21 event Protocol failure -- Unicall/21 protocol error. Cause 32773 Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 Channel gains Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 Channel switching -- Hungup 'UniCall/21-1' == Spawn extension (from_merlin, 87, 1) exited non-zero on 'UniCall/55-1' Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Channel gains Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Channel switching Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Call control(6) Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Drop call(cause=Normal Clearing [16]) Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 1101 - [1/ 400/Answer/Accepted Paid] -- Hungup 'UniCall/55-1' Jun 9 15:19:42 WARNING[19390]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 - 0101 [1/ 1/Idle /Idle] Jun 9 15:19:42 WARNING[19390]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 R2 prot. err. [1/ 1/Idle /Idle] cause 32773 - Unexpected CAS bit pattern Jun 9 15:19:42 WARNING[19390]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 1001 - [1/ 1/Idle /Idle] Jun 9 15:19:42 WARNING[19390]: chan_unicall.c:2644 handle_uc_event: Unicall/21 event Protocol failure -- Unicall/21 protocol error. Cause 32773 Jun 9 15:19:42 WARNING[19390]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 - 1001 [1/ 1/Idle /Idle] Jun 9 15:19:42 WARNING[19390]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 1001 - [1/ 1/Idle /Idle] Many thanks and best regards, Peter. ___ --Bandwidth 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] My Kernel
Hi List; I did yum install kernel and yum install kernel-devel, now when I type 'uname' -a I have the following: [EMAIL PROTECTED] /]# 'uname' -a Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1 SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386 GNU/Linux And when I type rpm -q kernel, then I have the followig: [EMAIL PROTECTED] /]# rpm - q kernel kernel-2.6.20-1.2319.fc5 So the question now is: what is my kernel that my system is using it? And how I can make my system use the latest updated kernel? Regards Bilal Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php ___ --Bandwidth 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] My Kernel
bilal ghayyad wrote: Hi List; I did yum install kernel and yum install kernel-devel, now when I type 'uname' -a I have the following: [EMAIL PROTECTED] /]# 'uname' -a Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1 SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386 GNU/Linux And when I type rpm -q kernel, then I have the followig: [EMAIL PROTECTED] /]# rpm - q kernel kernel-2.6.20-1.2319.fc5 So the question now is: what is my kernel that my system is using it? And how I can make my system use the latest updated kernel? Regards Bilal not to be rude, but what does this have to do with asterisk? From what you are telling us, I guess you need to find some fedora or general linux support medium... Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth 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] Crashes with Spandsp, app_rxfax.c, and asterisk 1.4.4
On Mon, Jun 11, 2007 at 11:21:49PM -0500, Rob Ristroph wrote: Hi everybody, I have a Fedora Core 4 x86 32 bit install, which I recently upgraded from asterisk 1.2 to the office 1.4.4 tarball. In the process of doing that I had to upgrade some autoconf/automake stuff, but it worked fine, and my new asterisk works fine. Except that anytime I receive a fax with spandsp and app_rxfax, asterisk seg faults. I have applied the spandsp patch of course, and I used the newer app_rxfax.c and app_txfax.c from soft-switch.org for 1.4. I have tried numerous versions of spandsp in turn, recompiling the rxfax and txfax application after installing each one. In particular, I am still getting this when using the latest spandsp snapshot from June 8th. http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/ Do you refer to those files? The patch there patches configure.ac . You need to run: ./bootstrap.sh ./configure I my searching, I found a few other people who mentioned the same problem, but they either didn't say if they solved it or were vague about how they did. My verison of libtiff is 3.7.1. One of the crashes printed out this stack trace information, although that doesn't come out most of the time: XXX -- Executing [EMAIL PROTECTED]:2] Set(Zap/1-1, [EMAIL PROTECTED]) in new stack -- Executing [EMAIL PROTECTED]:3] RxFAX(Zap/1-1, /var/spool/asterisk/fax/1181537898.0.tif|debug) in new stack linux40*CLI *** glibc detected *** /usr/sbin/asterisk: free(): invalid next size (normal): 0x09d5a908 *** Can you point gdb to your source tree so you'll have meaningful symbols there? === Backtrace: = /lib/libc.so.6[0x5471e0] /lib/libc.so.6(__libc_free+0x77)[0x54772b] /usr/lib/asterisk/modules/app_rxfax.so[0xebbc41] /usr/sbin/asterisk[0x80c3cc8] /usr/lib/asterisk/modules/app_macro.so[0xa330ea] /usr/sbin/asterisk[0x80c3cc8] /usr/sbin/asterisk[0x80c5002] /usr/sbin/asterisk[0x80c5d3e] /usr/sbin/asterisk[0x80f1f99] /lib/libpthread.so.0[0x656bd4] /lib/libc.so.6(__clone+0x5e)[0x5ae4fe] === Memory map: 00111000-00118000 r-xp 03:01 5330798 /usr/lib/asterisk/modules/res_musiconhold.so 00118000-00119000 rwxp 7000 03:01 5330798 /usr/lib/asterisk/modules/res_musiconhold.so 00119000-0011f000 r-xp 03:01 5330792 /usr/lib/asterisk/modules/res_config_pgsql.so 0011f000-0012 rwxp 5000 03:01 5330792 /usr/lib/asterisk/modules/res_config_pgsql.so 0012-00218000 r-xp 03:01 196501 /opt/lumenvox/engine_6.5/lib/libcrypto.so.0.9.7f 00218000-0022a000 rwxp 000f8000 03:01 196501 /opt/lumenvox/engine_6.5/lib/libcrypto.so.0.9.7f 0022a000-0022d000 rwxp 0022a000 00:00 0 X The memory map continues on for several pages but I can supply it if anyone thinks it would be useful. Most of the time the *CLI prompt just shows asterisk disconnecting, and the /var/log/asterisk/full just shows RxFax as the last thing ran, and the message of asterisk starting up again, like this: X [Jun 10 18:26:12] VERBOSE[13094] logger.c: -- Executing [EMAIL PROTECTED]:2] Set(Za p/1-1, [EMAIL PROTECTED]) in new stack [Jun 10 18:26:12] DEBUG[13094] app_macro.c: Executed application: Set [Jun 10 18:26:12] VERBOSE[13094] logger.c: -- Executing [EMAIL PROTECTED]:3] RxFAX( Zap/1-1, /var/spool/asterisk/fax/1181517967.7.tif|debug) in new stack [Jun 10 18:26:17] VERBOSE[13163] logger.c: Asterisk Event Logger Started /var/log/asterisk /event_log [Jun 10 18:26:17] VERBOSE[13163] logger.c: Asterisk Dynamic Loader Starting: [Jun 10 18:26:17] VERBOSE[13163] logger.c: == Parsing '/etc/asterisk/modules.conf': [Jun 10 18:26:17] VERBOSE[13163] logger.c: Found X Is anyone else out there seeing this ? Does anyone have any suggestions, even if it is only how to get more debuging information out ? From the reading and searching I have been doing the last several hours, it appears that the best thing to do in the long run might be to install iaxmodem and HylaFax. However, right now I have my asterisk invoking some custom scripts and uploading the faxes into a database based on DID and CallerID numbers, and I would much rather get this working now this way instead of have to learn how to integrate the same thing with HylaFax. Thanks in advance, --Rob -- http://rgr.freeshell.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 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL
Re: [asterisk-users] My Kernel
On Thu, Jun 14, 2007 at 03:02:20PM -0700, bilal ghayyad wrote: Hi List; I did yum install kernel and yum install kernel-devel, now when I type 'uname' -a I have the following: [EMAIL PROTECTED] /]# 'uname' -a Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1 SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386 GNU/Linux Original kernel of FC5, I believe. And when I type rpm -q kernel, then I have the followig: [EMAIL PROTECTED] /]# rpm - q kernel kernel-2.6.20-1.2319.fc5 That's a far more decent kernel version. So the question now is: what is my kernel that my system is using it? And how I can make my system use the latest updated kernel? You have installed a new kernel but have not yet booted your system from it. Your system is using 2.6.15-1.2054_FC5smp . (For newer versions I expect there to be no separate smp kernel version as the default kernel will now support SMP with no unnecessary overhead) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth 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] My Kernel
On 6/14/07, Remco Post [EMAIL PROTECTED] wrote: bilal ghayyad wrote: Hi List; I did yum install kernel and yum install kernel-devel, now when I type 'uname' -a I have the following: [EMAIL PROTECTED] /]# 'uname' -a Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1 SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386 GNU/Linux And when I type rpm -q kernel, then I have the followig: [EMAIL PROTECTED] /]# rpm - q kernel kernel-2.6.20-1.2319.fc5 So the question now is: what is my kernel that my system is using it? And how I can make my system use the latest updated kernel? Regards Bilal not to be rude, but what does this have to do with asterisk? From what you are telling us, I guess you need to find some fedora or general linux support medium... That's true. But isn't it easier to tell him to check his /boot/grub/grub.conf file? And only one line... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Unicall + MFC/R2 line dropped immediately afterconnect
Hi, Clearback signal due to billing pulses normally drops calls after a fixed amount of time 2 minutes or so, Can you stablish an outbound call and after a while it drops? Or it never succeds? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Peter Gubis Enviado el: jueves, 14 de junio de 2007 18:50 Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Unicall + MFC/R2 line dropped immediately afterconnect Hi, I am trying to set up an E1 line with CAS signaling using available unicall patches with libmfcr2 implementation. Inbound calls works well, I am able to get DNIS and ANI from incoming call, but I am still not able to make an outbound call with our local carrier. After tweaking of protocolvariant parameter in unicall.conf I was able to find proper values for inbound and outbound calls, but when outbound connection is created, it is immediately dropped by opposite side immediately after connection is made. I think, that we're receiving billing pulses on the line and they're interpreted as a cleaning request. I found similar problem in the list before: http://lists.digium.com/pipermail/asterisk-users/2006-June/156732.html. Steve suggested to use some kind of timeout to workaround this problem. I've tried to change many timeouts in libmfcr2 library, without any success. Can somebody help me to go around this line drop-down problem? Relevant part of debug log file: Jun 9 15:19:27 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 - 6 off [2/ 40/Group I /DNIS] Jun 9 15:19:27 WARNING[19398]: chan_unicall.c:2644 handle_uc_event: Unicall/21 event Alerting Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 - 0101 [1/ 200/Await answer /DNIS] Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:2644 handle_uc_event: Unicall/21 event Connected -- UniCall/21-1 answered UniCall/55-1 Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:1411 unicall_answer: Answer Call Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Call control(5) Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Answer call Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 0101 - [1/ 20/Group B /Accepted Paid] -- Attempting native bridge of UniCall/55-1 and UniCall/21-1 Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:1620 unicall_bridge: unicall_bridge called Jun 9 15:19:41 WARNING[19398]: chan_unicall.c:2644 handle_uc_event: Unicall/55 event Answered Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 - 0001 [1/ 400/Answered /DNIS] Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 R2 prot. err. [1/ 400/Answered /DNIS] cause 32773 - Unexpected CAS bit pattern Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 1001 - [1/ 1/Idle /Idle] Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:2644 handle_uc_event: Unicall/21 event Protocol failure -- Unicall/21 protocol error. Cause 32773 Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 Channel gains Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 Channel switching -- Hungup 'UniCall/21-1' == Spawn extension (from_merlin, 87, 1) exited non-zero on 'UniCall/55-1' Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Channel gains Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Channel switching Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Call control(6) Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Drop call(cause=Normal Clearing [16]) Jun 9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 1101 - [1/ 400/Answer/Accepted Paid] -- Hungup 'UniCall/55-1' Jun 9 15:19:42 WARNING[19390]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 - 0101 [1/ 1/Idle /Idle] Jun 9 15:19:42 WARNING[19390]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 R2 prot. err. [1/ 1/Idle /Idle] cause 32773 - Unexpected CAS bit pattern Jun 9 15:19:42 WARNING[19390]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 1001 - [1/ 1/Idle /Idle] Jun 9 15:19:42 WARNING[19390]: chan_unicall.c:2644 handle_uc_event: Unicall/21 event Protocol failure -- Unicall/21 protocol error. Cause 32773 Jun 9 15:19:42 WARNING[19390]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 - 1001 [1/ 1/Idle /Idle] Jun 9 15:19:42 WARNING[19390]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 1001 - [1/ 1/Idle /Idle] Many thanks and best regards, Peter.
Re: [asterisk-users] ODBC voicemail questions
On 6/14/07, Kyle Sexton [EMAIL PROTECTED] wrote: Hey Kyle! 1. Is the voicemail message stored in the datbase, or just it's location/filename? Yes, the voicemail message itself is stored in the database, as a BLOB or large object file. 2. Does MWI propagate when using an ODBC backend? Yes. 3. If it does both of those things, wouldn't it work well for a centralized voicemail system instead of a solution like NFS or weird rsync scripts? Yes, that's the whole reason it was written :-) -Jared ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending text to a phone that's no in-use ...
Hi! Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, Hello World) but that doesn't exist ... Snom's or Grandstream GXP2000's I'm afraid... Sending text to them while in a call works fine (although reading the text on the Snom was challenging until I'd worked out I needed to reprogram one of the function keys :) Read the part about .call file here: http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText Or look at the FAQ at the bottom of this page, look for sipsak: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+snom Concept: Configure an auto-answer line on your phone and use a .call file together with SendText(), or use sipsak instead (i.e. external tool, outside of Asterisk). Third option if you have a SNOM 360 or 370: Send a SIP NOTIFY and push a XML message along with that (mini-browser style). Read more about this on the SNOM knowledgebase/wiki. Cheers, Philipp ___ --Bandwidth 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] Que on A2Billing
That was easy... Thanks a million man... Dunno what I was thinking and went too far writing custom scripts... Cheers, Nitesh Guillermo Salas M. wrote: On Thu, 2007-06-14 at 14:46 -0400, Nitesh Divecha wrote: Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the IVR Prompt... Meaning from my XLite dialer I want to dial directly and let A2Billing do the billing part. Right now is something like when I dial any number from XLite, A2Billing script is invoked and it will announce You have XXX amount, please enter the number you wish to call followed by #. And then I have to enter the number again and then the call is initiated... Its kinda annoying to do that every time you want to call. Is there anyway to modify config some where, so it will do the billing in background when the phone call is hangup. Yes, is possible using the a2billing.conf file in the right way. I don't have the v1.3 installed, but in the previous release 1.2.3 you must have to modify : use_dnid=YES number_try=1 say_balance_after_auth=NO say_balance_after_call=NO say_rateinitial=NO say_timetocall=NO Regards, ___ --Bandwidth 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] testing
Please disregard. ___ --Bandwidth 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] My Kernel
David Gomillion wrote: That's true. But isn't it easier to tell him to check his /boot/grub/grub.conf file? And only one line... Easier, but not smarter. If you'll excuse me I have some wild animals to feed. -- Jeff Davis Netsource Consulting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement
On 6/14/07, Nick Seraphin [EMAIL PROTECTED] wrote: On Thu, 14 Jun 2007, C F wrote: On 6/13/07, Erik Anderson [EMAIL PROTECTED] wrote: On 6/13/07, C F [EMAIL PROTECTED] wrote: This is just weird I wrote it in caps so you can read it but you still didn't read it so here it is again: its a T1 card that does NOT have a CSU in it, and it is working fine and yes it is a T1 providing PRI. sarcasm Dang shmaltz. You've convinced us - we've all been wasting our precious money on CSUs this whole time. We're all idiots! /sarcasm Seriously - if you're so sure about your card not having a CSU, what is the make/model? Pony up, man. It's a Panasonic KX-TA0187 for T1, or KX-TA02290 The docs and technicians say it doesn have one AND that the FCC requires it. Hence my qeustion does the FCC require it. I think what he's referring to is really the KX-TD187... which is a T1 interface module for the Panasonic KX-TD1232 Digital Hybrid Phone System (I have one of these systems, but not the T1 module). No, I am reffering to KX-TA0187 which is a T1 card for the Panaosnic KX-TDA line. and the KX-TA0290 which is a PRI card for the Panasonic KX-TDA line. Now there is a KX-TA1232 analog system, and maybe there was a KX-TA187 module for it that has since been discontinued... but I think he meant the digital one. http://www.ablecomm.com/t1isdideq.html They do SAY it doesn't have a CSU... but it's beyond my understanding of how it could possibly work without one. They seem to sell a separate CSU module that can go with it. Maybe he's just not seeing the extra little box because there's more wire between that and the demarc? Was this a system that was already installed for you? Or did you install it yourself? Maybe the CSU is external and you just didn't recognize/see it there? No, I installed the system, it goes from smart jack to the PRI card in the Panasonic KX-TDA200 thru the 0290 card When I first started working with T1's, most CSU's were external. I still have several of them in storage in fact... and I still use external CSU/DSU's on my production network today. :-) I'm typing this message and it will be sent over a T1 connected to 2 external CSU's before it reaches the internet. Bottom line is, no matter what the FCC says... and if somehow you managed to get it to work without a CSU... I believe the phone company would have a fit if they knew you connected equipment to their network without a CSU on it. They're very big on standards-compliance and stuff like that. Sometime look into their rules and regs about colocating equipment inside one of their CO's... it's very very strict. The last thing you say is why I am asking this question. The compliance doesn't realy bother me that much, what I am afraid is if the provider notices this and decides to cut it because of that. -- Nick ___ --Bandwidth 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] Slightly OT:CSU on Digium cards, and it's requirement
What I am thinking is that a CSU could provide mutiple functions, error handling, diagnostics and signal boosting, which is not built into the Panasonic equipment, but the lower level signaling that a CSU could provide is built into it, and that's why it works. As far as I knew before I read it was that it shoudn't work, but according to that documentation it is suppose to work. On 6/14/07, C F [EMAIL PROTECTED] wrote: On 6/14/07, Nick Seraphin [EMAIL PROTECTED] wrote: On Thu, 14 Jun 2007, C F wrote: On 6/13/07, Erik Anderson [EMAIL PROTECTED] wrote: On 6/13/07, C F [EMAIL PROTECTED] wrote: This is just weird I wrote it in caps so you can read it but you still didn't read it so here it is again: its a T1 card that does NOT have a CSU in it, and it is working fine and yes it is a T1 providing PRI. sarcasm Dang shmaltz. You've convinced us - we've all been wasting our precious money on CSUs this whole time. We're all idiots! /sarcasm Seriously - if you're so sure about your card not having a CSU, what is the make/model? Pony up, man. It's a Panasonic KX-TA0187 for T1, or KX-TA02290 The docs and technicians say it doesn have one AND that the FCC requires it. Hence my qeustion does the FCC require it. I think what he's referring to is really the KX-TD187... which is a T1 interface module for the Panasonic KX-TD1232 Digital Hybrid Phone System (I have one of these systems, but not the T1 module). No, I am reffering to KX-TA0187 which is a T1 card for the Panaosnic KX-TDA line. and the KX-TA0290 which is a PRI card for the Panasonic KX-TDA line. Now there is a KX-TA1232 analog system, and maybe there was a KX-TA187 module for it that has since been discontinued... but I think he meant the digital one. http://www.ablecomm.com/t1isdideq.html They do SAY it doesn't have a CSU... but it's beyond my understanding of how it could possibly work without one. They seem to sell a separate CSU module that can go with it. Maybe he's just not seeing the extra little box because there's more wire between that and the demarc? Was this a system that was already installed for you? Or did you install it yourself? Maybe the CSU is external and you just didn't recognize/see it there? No, I installed the system, it goes from smart jack to the PRI card in the Panasonic KX-TDA200 thru the 0290 card When I first started working with T1's, most CSU's were external. I still have several of them in storage in fact... and I still use external CSU/DSU's on my production network today. :-) I'm typing this message and it will be sent over a T1 connected to 2 external CSU's before it reaches the internet. Bottom line is, no matter what the FCC says... and if somehow you managed to get it to work without a CSU... I believe the phone company would have a fit if they knew you connected equipment to their network without a CSU on it. They're very big on standards-compliance and stuff like that. Sometime look into their rules and regs about colocating equipment inside one of their CO's... it's very very strict. The last thing you say is why I am asking this question. The compliance doesn't realy bother me that much, what I am afraid is if the provider notices this and decides to cut it because of that. -- Nick ___ --Bandwidth 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] Slightly OT:CSU on Digium cards, and it's requirement
On Thu, 14 Jun 2007, C F wrote: but the lower level signaling that a CSU could provide is built into it Possible. In any event, it is this function that describes the essential aspects of a CSU. But I think the standard is very clear on the requirements for OAMP stuff too. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth 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] Slightly OT:CSU on Digium cards, and it's requirement
On Thu, 14 Jun 2007, C F wrote: Bottom line is, no matter what the FCC says... and if somehow you managed to get it to work without a CSU... I believe the phone company would have a fit if they knew you connected equipment to their network without a CSU on it. They're very big on standards-compliance and stuff like that. Sometime look into their rules and regs about colocating equipment inside one of their CO's... it's very very strict. The last thing you say is why I am asking this question. The compliance doesn't realy bother me that much, what I am afraid is if the provider notices this and decides to cut it because of that. Whether they would actually cut you off or not probably depends on A) if they find out about it, and B) whoever finds out about it is a strict play-by-the-rules kinda guy and/or has a grudge against you or is having a bad day. A lot of telco employees tend to look the other way... especially if it's not their job to care about it. But... they would have every right to terminate the service if you don't have proper equipment connected to their network. So if they DID decide to terminate it, they would legally have the right to do so, and you would have no recourse other than possibly to purchase the correct equipment and maybe pay a reconnect fee to get service turned back on, which may take days/weeks/whatever time frame to do so. So it's basically a question of, can you afford the downtime caused by them shutting you off if/when they ever found out and/or cared enough to follow the rules. The other possibility, considering it is working for you now, is that there IS a CSU built in but they don't want to tell you... maybe for example because it's not FCC certified... or so that they can charge you for an external CSU. -- Nick ___ --Bandwidth 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] Reinvite / one-way media.
I have two phones on a network behind NAT. Enabling canreinvite=yes on the Asterisk server allows them to talk to each other very effectively through the local network. Unfortunately, calling any outside destinations yields one-way media issues where the far end can hear me but I can't hear them, probably due to lack of an ALG on the NAT router that understands the SDP negotiation of the RTP ports. I like reinvites in theory--it leads to far more efficient and low-latency interaction with my outside termination providers by saving at least one round trip up and down the coast, which is the result of where the Asterisk server is in relation to the phones and the termination provider. What can I do? I suppose one approach is to hard-pin and forward the RTP ports both phones negotiate to; both phones have that capability. Then I can forward them (DNAT) on the gateway. Anything easier or rather obvious I'm missing? Thanks! -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth 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] app_rxfax vs (iaxmodem+hylafax)
can anybody help me to choose the most reliable fax solution for * . after googling the net i found that there are at least two solutions for this, app_rxfax+spandsp and iaxmodem+hylafax. - what's the differences between these two? - which one's better? why? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
