Re: [Asterisk-Users] SPA-3000 and incoming faxes
If you are willing to dedicate a fax line and forward faxes to a dedicated extension it work 100% for Incoming and Outgoing faxes with Sipura-3000 Though, you have to change in the Regional Tab: Ring Waveform: from Sinusoid to Trapezoid I've been using NVbackgroundDetect with Sipura-3000 and forwarding the to fax extension (Hylafax) if it is a fax and voice line if it is a voice call. So far I've been receiving faxes from Europe, Asia, USA and it work 98% most of the time. I have only one customer in Mexico and one in Asia that NVbackgroundDetect has a problem with to recognize fax signal. -- #Joseph On Mon, 2005-09-26 at 23:08 -0500, Tim Litwiller wrote: I've been running with a generic X100P for 5 or so months and every once in a while I have problem receiving faxes. I see that others have the same problems and some worse than I have with these boards so I was wondering if using a Sipura SPA-3000 would be any more reliable. Has anyone had enough experience to tell me if that would definitely fix the random fax error. PS. I have * at home and have it configured to send faxes to an extension that has a fax machine connected to it and the fax machine is set to auto answer the first ring. It is connected to a SPA-2002 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bad FCS nightmare to Nortel SL100 with TE410P
I have an * box connected to a Nortel SL100 through a PRI (US) using the Digium TE410P (quad-span T1 card). I don't have access to the SL100 - it is handled by another group. The span comes up OK (timing, framing fine). However, as soon as the D channel comes up, I get endless HDLC Bad FCS errors. I modified logger.conf to get rid of the messages (so I could see what else was going on), and noticed that the B-channel restart was going horribly slow, and the D channel was essentially flapping up and down. I could sometimes squeeze a call in while the D channel was up, but it would only last a few seconds. I also get short write errors as well (unfortunately I don't have a log of these and can't get at the PRI at the moment to get the exact error message). I've had the physical circuit tested and there are no issues with it. In fact, it was working fine to the same switch as an EM digital trunk up until we tried to change it to a PRI. I've tried 3 different TE410Ps on three different * versions (based on things I've seen in previous posts). All behave exactly the same. The versions are 1.0.5, 1.0.9, and a CVS version of 1.2.0-beta1 pulled down at the end of August. In all cases, the systems are Dell PowerEdge 1750s (using RAID, no IDE drives involved) on Debian / kernel 2.4.27. I see no indication of problematic interrupts. In one test, there were 3 other PRIs running on the TE410P (in production) and there are no problems with any other PRIs. Ditto the configuration (I've checked and am doing the exact same thing with all my PRIs, just on different channels). Before I start providing configuration excerpts - has anyone had this problem connecting to an older Nortel Meridian switch and if so, what did you do to fix it? I suspect that there is a subtle configuration option on the SL100 that is wrong, but since I don't have access to it I can't confirm that. Can the wrong switch type cause FCS errors? Is there anything specific I can look at? For those who speak SL100, do you know of any specific parameter I can point the SL100 guy to? One more data point: I threw the PRI from the SL100 onto a spare port on a Cisco AS5350 and the AS5350 isn't complaining (no frame slips, no problem with the D channel). I'm pulling my hair out with this. Any help or pointers to info would be helpful. I will post a summary to the list if I get any useful private e-mail about this. Thanks! --- Gil Kloepfer [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Non-blocking Dial (and other commands): is there a way?
In order to use a with GrandStream BT-488 as pass-through gateway, I need a way of sending the FXO port off hook when I'm using the FXS port for VoIP communications, because I want to use the hunting line feature to let incoming call skip that FXO port and move on to the next free line. The only way I have found to engage a device without getting blocked until the call ends passes through an AGI script that drops a callfile into the /var/spool/asterisk/outgoing directory, telling Asterisk to dial the FXO port and then connect the channel to, say, the MusicOnHold() application. When I'm done, I can then issue a SoftHanghup() to the FXO device. This method strikes me as pretty clumsy: aren't there better ways of issuing commands from the dialplan in detached mode, perhaps getting a handle useful to regain control later, and proceed to do other things? Enzo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IBM x306 - some progress
Hi Marco, As far as I can recall, the IBM setup utility can enable you to change the IRQ of the SCSI controller. In addition, I've never seen a WildCard board bound to IRQ7 on any box, which is very weird in it self. I'm flying over to Ireland today (actually, at the airport right now), and I'm coming back on Sunday. If You'd like, you can bring your box to my office after Rosh-Hashana, and I'll try to help you out. Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Supino Sent: Monday, September 26, 2005 10:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IBM x306 - some progress Hi, I asked yesterday about a problem with x306 and IRQ sharing, didnt get much info, now, i was playing with lspci, and see something strange, lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is also on IRQ 7, lspci -bv (from the man - b - shows bus-centric view, as seen by the BUS and not by the kernel) shows me the TDM400P is on IRQ 5, why does the kernel puts it on IRQ 7 ? any insights much appriciated. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream 496 not working on cordless phone
On Mon, 2005-09-26 at 17:24 -0400, Nana Tandoh wrote: Hi All, We are using SER/Asterisk, it works fine from X-lite to corded phones but have problems using a cordless phone on the Handytone 496. Has anyone experienced this problem. Well, if you told us what the problems are perhaps we could help. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What ISDN hardware would you recommend?
On Mon, 26 Sep 2005, Francesco Peeters wrote: Trying again... *Summary:* I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode; What card(s) should I put in to these servers? *The long story:* I have 3 locations I want to connect using (*) servers. 1 of those has a single BRI with a Siemens DECT PABX. 1 of those has two BRI's with 2 Siemens DECT PABX's, each serving a different area. 1 of those has two BRI's and a 2 port Nova Compact PABX with DECT First step would be to set up the (*) servers and have them interconnected. When all of that works we'd go on to connect them to the ISDN and connect the existing PABX's to the servers so we can - for now - maintain the existing environment but use (*) to route traffic on a least cost basis, as well as allow SIP/IAX connections from out of office locations. The machines themselves will not pose much of a problem, but what ISDN hardware would you recommend for this? (1 site with 1 TE and 1 NT mode port, 2 sites with 2 TE and 2 NT mode ports) I cannot compare with other cards, but I recommend the Eicon DIVA Server Cards (like 4BRI). I only use them and they work very good. Armin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip, call ransfer and call waiting
trixter http://www.0xdecafbad.com a écrit : On Mon, 2005-09-26 at 11:08 +0200, Daniel ANDRE wrote: Hello all, I have a very basic question but I haven't found any answer. I would like to configure asterisk so that it wil not indicate a call waiting to a SIP phone if it is already on conversation (off hook). But I don't want to loose call transfer, call hold and so on. Is there any possibility to do that? Yup... exten = 123,1,SetGroup(user1) exten = 123,2,CheckGroup(1) ; dont let more than 1 call at a time exten = 123,3,Dial(sip/user1) exten = 123,103,Busy ; this is where it goes if CheckGroup indicates more than X calls ... see http://voip-info.org/wiki-Asterisk+cmd+SetGroup for more info. You may have to play games with variables to make a macro perhaps that would be more generic in this regard, but this should at least get you started. Thank you for this pointer. I have seen that tere is a but in current stable (http://bugs.digium.com/bug_view_page.php?bug_id=0003067). In the bug report there is a reference to group categories. If I don't use categories do I need the patch? Regards, Daniel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Problem setting up TDM22B card
Hi, I did dmesg | tail it says ... dmesg | tail f6 != 58 f7 != 59 f8 != 58 f9 != 59 fa != 58 fb != 59 fc != 58 fd != 59 fe != 58 Freshmaker failed register test This is small part of dmesg output which may help in diagnosing the problem. Registered Tormenta2 PCI No ISA tormenta card found at d usb.c: registered new driver wcusb Wildcard USB FXS Interface driver registered Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) Freshmaker version: 58 00 != 18 01 != 19 02 != 18 03 != 19 04 != 18 05 != 19 06 != 18 07 != 19 08 != 18 09 != 19 0a != 18 0b != 19 . (it goes on like this which I don't understand!) f9 != 59 fa != 58 fb != 59 fc != 58 fd != 59 fe != 58 Freshmaker failed register test Can we make out something from this? Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Sep 26, 2005 at 10:00:10PM -0400, Patrick wrote: I know it seems basic but did you make sure and plug power into the board when you installed it into the PCI slot? I spent about three hours trying to get the dang thing to work in my machine until I decided to stick the card into another PCI slot. That is when I noticed that I had forgotten to ALSO plug power into the board from the power supply. Everything worked fine after that (yep, I was a noob). :-) You get a messge about it from the module at module load time. rmmod wctdm (or wcfxo) and re- modprobe it, and then run: dmesg |tail -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Problem setting up TDM22B card
Hi, I know some bits about zaptel.conf and zapata.conf but problem is modules wctdm, wcfxo, wcfxs are not getting loaded. modprobe zaptel is successful! Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 25, 2005 at 11:50:06PM -0700, somesh s wrote: Hi, I got the following output when I run the command. genzaptelconf -svd ./genzaptelconf: line 616: /etc/init.d/asterisk: No The script assumes that there is an /etc/init.d/asterisk script. Please stop asterisk manually. such file or directory Unloading zaptel modules: wcusb zaptel Test Loading modules: - zaptel - zaphfc - qozap Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wctdm Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcfxo Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcfxs - pciradio Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - tor2 Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - torisa Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wct1xxp Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wct4xxp Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcte11xp - wcusb - ztd_eth It tries and fails to load many modules. This is expected, as you don't have their hardware. However can you load any zaptel module using modprobe? What's the output of: 'lsmod | grep zaptel' BTW: if you have a good idea of what should be in zaptel.conf and zapata.conf, I figure you should write them manually and not waste too much time on that automated script. It is meant to save time, not to spend time. Updating '/etc/default/zaptel' Generating '/etc/zaptel.conf' Generating '/etc/asterisk/zapata-channels.conf' Reconfiguring identified channels Zaptel Configuration == Channel map: 0 channels configured. ./genzaptelconf: line 653: /etc/init.d/asterisk: No such file or directory Checking channels configured in Asterisk: ./genzaptelconf: line 665: asterisk: command not found What may be the problem? Help me in this regard. Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 25, 2005 at 10:42:52PM -0700, somesh s wrote: Hi, Can you please give me some details about the link you have sent? I am not aware of what it does? [http://tzafrir.org.il/genzaptelconf] It is a bash script for generating zaptel.conf and zapata.conf according to the current settings. To use it: wget http://tzafrir.org.il/genzaptelconf bash genzaptelconf -h # gives help Try -s and -v . -d is probably not recommended if you have more thn one card, I figure. Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Sep 23, 2005 at 06:22:06AM -0700, somesh s wrote: Hi Steve, This is zaptel.conf. Can you please tell me if you require to see more conf files? [zaptel.conf] loadzone = us defaultzone=us fxoks=1-2 fxsks=3-4 http://tzafrir.org.il/genzaptelconf Should auto-detect zaptel.conf settings. Just in case you're not sure. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | === message truncated === __ Yahoo! for Good Donate to the Hurricane Katrina relief effort. http://store.yahoo.com/redcross-donate3/
Re: [Asterisk-Users] Re: Problem setting up TDM22B card
Hi, And I have the digium's hardware too in one of my PCI slots. Regards, Somesh S. Shanbhag --- somesh s [EMAIL PROTECTED] wrote: Hi, I know some bits about zaptel.conf and zapata.conf but problem is modules wctdm, wcfxo, wcfxs are not getting loaded. modprobe zaptel is successful! Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 25, 2005 at 11:50:06PM -0700, somesh s wrote: Hi, I got the following output when I run the command. genzaptelconf -svd ./genzaptelconf: line 616: /etc/init.d/asterisk: No The script assumes that there is an /etc/init.d/asterisk script. Please stop asterisk manually. such file or directory Unloading zaptel modules: wcusb zaptel Test Loading modules: - zaptel - zaphfc - qozap Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wctdm Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcfxo Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcfxs - pciradio Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - tor2 Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - torisa Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wct1xxp Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wct4xxp Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcte11xp - wcusb - ztd_eth It tries and fails to load many modules. This is expected, as you don't have their hardware. However can you load any zaptel module using modprobe? What's the output of: 'lsmod | grep zaptel' BTW: if you have a good idea of what should be in zaptel.conf and zapata.conf, I figure you should write them manually and not waste too much time on that automated script. It is meant to save time, not to spend time. Updating '/etc/default/zaptel' Generating '/etc/zaptel.conf' Generating '/etc/asterisk/zapata-channels.conf' Reconfiguring identified channels Zaptel Configuration == Channel map: 0 channels configured. ./genzaptelconf: line 653: /etc/init.d/asterisk: No such file or directory Checking channels configured in Asterisk: ./genzaptelconf: line 665: asterisk: command not found What may be the problem? Help me in this regard. Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 25, 2005 at 10:42:52PM -0700, somesh s wrote: Hi, Can you please give me some details about the link you have sent? I am not aware of what it does? [http://tzafrir.org.il/genzaptelconf] It is a bash script for generating zaptel.conf and zapata.conf according to the current settings. To use it: wget http://tzafrir.org.il/genzaptelconf bash genzaptelconf -h # gives help Try -s and -v . -d is probably not recommended if you have more thn one card, I figure. Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Sep 23, 2005 at 06:22:06AM -0700, somesh s wrote: Hi Steve, === message truncated === __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
[Asterisk-Users] Termcap missing (compile error [editline/libedit.a] Error 1)
Hello Gentlemen :-) I am a little disapointed by an error occured during an update from 1.0.7 to Head in a Debian testing distro. The first error message happens by using the famous script from http://www.szmidt.org/asterisk/asterisk-update.sh : configure: error: termcap support not found make: *** [editline/libedit.a] Erreur 1 ERROR! Compile exited with error. Aborting script! And, if I tempt to compile manualy with make clean; make; make install, I can see that at the end : cd editline unset CFLAGS LIBS test -f config.h || ./configure loading cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc ) works... yes checking whether the C compiler (gcc ) is a cross-compiler... no checking whether we are using GNU C... yes checking whether gcc accepts -g... yes checking how to run the C preprocessor... gcc -E checking host system type... i686-pc-linux-gnu cygwin detected checking for a BSD compatible install... install checking for ranlib... ranlib checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/libedit.a] Erreur 1 sarge:/usr/src/asterisk# What occurs ? What I have missed ? Any idea to help me ? What can I describe or search more for a best analyze ? Many thanks in advance, guys ! Best Regards, Francois BERGERET, France. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Removing - (Dash) from Dialed Numbers
[EMAIL PROTECTED] wrote:
I am trying to enable dial-by-email by using LDAPget to query
an Active Directory server. I've got it retrieving the phone
number fine. Unforunately, the numbers stored in active
directory are either in the format: (xxx) xxx- or
xxx-xxx-.
Is there
any way to parse characters out of the dialed phone number so
that I only end up with digits (remove spaces, parenthesis and
dashes)? From there, my outbound routes can take care of
where to send the call.
This would be darned easy to do with the AGI and a perl script.
IE:
exten = _X.,1,agi,fixnumbers|${MyNumber}
exten = _X.,2,Dial(ZAP/g0/1${MyNumber})
Then, in a perl script called fixnumbers and inside the agi-bin
directory:
## START CODE #
#!/usr/bin/perl -w
use strict;
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $number=$ARGV[0];
$number=~s/-//g;
$number=~s/ //g;
$number=~s/\(//g;
$number=~s/\)//g;
print $AGI-set_variable('MyNumber',$number);
exit;
### END CODE
Depending on how many calls per second you want to perform,
some dialplan magic might be cheaper than starting up a
perl process.
I'd write a diaplan macro for this. If the numbers are in a
fixed format (4th character is a -, 7th character is a -,
etc), then it's really simple.
Something like this:
exten = s,1,SetVar(strPart1 = ${myNumber:0:3}
exten = s,2,SetVar(strPart2 = ${myNumber:4:3}
exten = s,3,SetVar(strPart3 = ${myNumber:7:3}
exten = s,4,SetVar(myNumber = $strPart1$strPart2$strPart3
But I'm using quite an old Asterisk, so current syntax might
be a little different, but the Wiki suggests this still works.
--
Andreas Sikkema bbned NV
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0)10 2245544 f: +31 (0)10 413 65 45
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Re: [Asterisk-Users] Termcap missing (compile error [editline/libedit.a] Error 1)
On Tue, Sep 27, 2005 at 09:20:13AM +0200, [EMAIL PROTECTED] wrote: Hello Gentlemen :-) I am a little disapointed by an error occured during an update from 1.0.7 to Head in a Debian testing distro. Start with defining a standard deb-src of Sarge (I think it is defined by default. Maybe remmed-out) and then run: apt-get install build-essential apt-get build-dep asterisk It should get you roughly the packages needed to build HEAD from source. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1)
You must install libncurses5-dev regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Enviado el: martes, 27 de septiembre de 2005 9:20 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1) Hello Gentlemen :-) I am a little disapointed by an error occured during an update from 1.0.7 to Head in a Debian testing distro. The first error message happens by using the famous script from http://www.szmidt.org/asterisk/asterisk-update.sh : configure: error: termcap support not found make: *** [editline/libedit.a] Erreur 1 ERROR! Compile exited with error. Aborting script! And, if I tempt to compile manualy with make clean; make; make install, I can see that at the end : cd editline unset CFLAGS LIBS test -f config.h || ./configure loading cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc ) works... yes checking whether the C compiler (gcc ) is a cross-compiler... no checking whether we are using GNU C... yes checking whether gcc accepts -g... yes checking how to run the C preprocessor... gcc -E checking host system type... i686-pc-linux-gnu cygwin detected checking for a BSD compatible install... install checking for ranlib... ranlib checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/libedit.a] Erreur 1 sarge:/usr/src/asterisk# What occurs ? What I have missed ? Any idea to help me ? What can I describe or search more for a best analyze ? Many thanks in advance, guys ! Best Regards, Francois BERGERET, France. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 23/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pbx_wilcalu.so: undefined symbol:
Anyone run into this? This is from the latest 1.2.0 beta1 tarball. Got it all compiled, but this undefined symbol is stopping asterisk from loading. Can I savely bypass this module and if so, what does it actually do? Cheers ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Early Media in 180 Ringing
Hi :) Am Montag, 26. September 2005 19:48 schrieb Ronald Voermans: Hello, As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content. How can this be solved? Well, I am not that expert but AFAIK your PSTN gateway should send a 183 (Session progress) than a simple 180. Do you use Dial(SIP/blah|30|m(moh_class)) to start early media? Regards, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Integration with NMS AG-E1/T1
I want to replace a custom PBX, that is infront on a IVR system based on OLD NMS AG-E1 Card. The Cards is configurated with CAS Digitalmode, someone can give me some info about Digim Cards CAS configuration i need a conversion Table? I wanto to don't touch configuration on winbox, i want only replace HWPBX box with asterisk. Diagram Telco E1 ===Proprietary PBX(CAS)===IVR Server AG-E1 Regards Configurations ag.cfg IdleCode = 0xD5, 0x5 DigitalMode= CAS ClockRef = NET1 _ Get free infected, boring, wrong, empty, or any other email for yourself. Go to --- http://www.mailchoose.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Initial release of AMPortal Debian/Xorcom-Rapidpackages
Tzafrir, I got these error messages installing AMP on your distribution rapid 1.1. Versions of Apache and PHP are the ones that come inside the package and nothing new has beeen added. Any idea about what went wrong? Regards, Jose M. Limeres /etc/apt apt-get install amportal Reading Package Lists... Done Building Dependency Tree... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. Since you only requested a single operation it is extremely likely that the package is simply not installable and a bug report against that package should be filed. The following information may help to resolve the situation: The following packages have unmet dependencies: amportal: Depends: asterisk-config-amportal (= 1.10.008-1) but it is not going to be installed Depends: amportal-common (= 1.10.008-1) but it is not going to be installed Depends: amportal-cdr (= 1.10.008-1) but it is not going to be installed Depends: amportal-vmail (= 1.10.008-1) but it is not going to be installed Depends: amportal-panel (= 1.10.008-1) but it is not going to be installed E: Broken packages -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Tzafrir Cohen Enviado el: 22 September 2005 15:41 Para: Asterisk Users list Asunto: [Asterisk-Users] Initial release of AMPortal Debian/Xorcom-Rapidpackages Hello Asteriskers, We are proud to announce, our initial support for AMPortal in Debian Sarge: we are releasing Debian packages containing AMPortal 1.10.008, partially working under our Rapid distribution, the packages should also run on normal Debian Sarge and hopefully also under Sid, Etch, Ubuntu or any other Debian variant without any problems (we hope ;) The packages are not really stable , so we do not recommend using them in production machines at this moment. We do encourage really brave men (or women) to install the packages and report any problems found, as this is still in beta/alpha stage. To install those packages you need to add these lines into /etc/apt/sources: # rapid amp repository deb http://rapid.dotsrc.org/ amportal/ deb-src http://rapid.dotsrc.org/ amportal/ and then execute: apt-get update apt-get install amportal If you do not have apache installed, this package should install apache1 (and the php4 packages needed), however this should also work with apache2 (untested at the moment). You should also note that those packages also modify /etc/php4/cli/php.ini and /etc/php4/apache/php.ini to include mysql.so, the user www-data is added to the asterisk group (otherwise you will not be able to modify /etc/asterisk/* from the browser), and some directories are chmod g+rw by asterisk. AMPortal is installed by default into /usr/share/amportal. You will need to expose it to the web-root of apache. If your webroot is in /var/www (Debian default), you have to run: cd /var/www ln -s /usr/share/amportal . There is a bug in /etc/amportal.conf configuration provided in those package, so you will also need to edit it manually. The line containing AMPWEBADDRESS=127.0.0.1 should be modified into contain your correct server name (I think localhost should be enough). Do not forget to run apply_conf.sh after that change: /usr/lib/amportal/apply_conf.sh You can find some other goodies on that dir (scripts for installing a clean database for example, an upgrade script). Next we we plan an upgrade to AMPortal 1.10.009, and then we plan on fixing all the things we broke on the package... (MOH, voicemail... etc). For more information, please see http://xorcom-rapid.berlios.de/ http://xorcom.com/ -- Tzafrir Cohen icq#16849755 +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Early Media in 180 Ringing
If guess I figured it out already. I made some changes in chan_sip.c (when ringing was received, it didn't check for SDP), and recompiled. It's working now! Ronald - -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Hauke Zuehl Verzonden: dinsdag 27 september 2005 10:02 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Early Media in 180 Ringing Hi :) Am Montag, 26. September 2005 19:48 schrieb Ronald Voermans: Hello, As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content. How can this be solved? Well, I am not that expert but AFAIK your PSTN gateway should send a 183 (Session progress) than a simple 180. Do you use Dial(SIP/blah|30|m(moh_class)) to start early media? Regards, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] failed make install on Solaris 10
I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found anywhere! make: *** [bininstall] Error 2 [EMAIL PROTECTED] # I would prefer not to install everything manually if possible. Anyone have any ideas how I get around this? Asterisk is clearly in the directory, but for some reason Solaris can't pick it up. Regards, Joe --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Sep 26, 2005 at 10:00:10PM -0400, Patrick wrote: I know it seems basic but did you make sure and plug power into the board when you installed it into the PCI slot? I spent about three hours trying to get the dang thing to work in my machine until I decided to stick the card into another PCI slot. That is when I noticed that I had forgotten to ALSO plug power into the board from the power supply. Everything worked fine after that (yep, I was a noob). :-) You get a messge about it from the module at module load time. rmmod wctdm (or wcfxo) and re- modprobe it, and then run: dmesg |tail -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -- Message: 19 Date: Tue, 27 Sep 2005 00:15:14 -0700 (PDT) From: somesh s [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Problem setting up TDM22B card To: Tzafrir Cohen [EMAIL PROTECTED] Cc: Asterisk Users asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi, I know some bits about zaptel.conf and zapata.conf but problem is modules wctdm, wcfxo, wcfxs are not getting loaded. modprobe zaptel is successful! Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 25, 2005 at 11:50:06PM -0700, somesh s wrote: Hi, I got the following output when I run the command. genzaptelconf -svd ./genzaptelconf: line 616: /etc/init.d/asterisk: No The script assumes that there is an /etc/init.d/asterisk script. Please stop asterisk manually. such file or directory Unloading zaptel modules: wcusb zaptel Test Loading modules: - zaptel - zaphfc - qozap Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wctdm Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcfxo Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcfxs - pciradio Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - tor2 Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - torisa Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wct1xxp Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wct4xxp Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcte11xp - wcusb - ztd_eth It tries and fails to load many modules. This is expected, as
Re: R: [Asterisk-Users] Problem setting up TDM22B card
Hi, I didn't get any solution in the mailing list. [http://asterisk.linkx.net/asteriskusers/200409/msg01167] What should be the next step? Changing the machine??? Is it machine dependent?... Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Sep 27, 2005 at 12:13:21AM -0700, somesh s wrote: Hi, I did dmesg | tail it says ... dmesg | tail f6 != 58 f7 != 59 f8 != 58 f9 != 59 fa != 58 fb != 59 fc != 58 fd != 59 fe != 58 Freshmaker failed register test I'd try Digium support. But google game me also: http://asterisk.linkx.net/asteriskusers/200409/msg01167.html I haven't bothred looking at the source yet. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Listening for DTMF when dialling
Hi all, I want to set up an extension which dials a group of phones while at the same time plays a message (Press 1 to leave a message) and listens for DTMF. I haven't played around yet but the way I read the docs this isn't possible as Thanks, Peter Spikings This message has been comprehensively scanned for viruses, please visit http://virus.e2e-filter.com/ for details. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Listening for DTMF when dialling (sorry, accidentally sent the previous message too early!)
Hi all, I want to set up an extension which dials a group of phones while at the same time plays a message (Press 1 to leave a message) and listens for DTMF. I haven't played around yet but the way I read the docs this isn't possible as the dial command doesn't have appropriate options and takes complete control of the channel. However surely this is a normal thing to want to do? Am I right thinking it's not possible? Are there any plans to have (say) a fork command which splits the channel into 2 or more threads (passing audio from the first specified to the caller) and another command (and option to dial) which make * abandon the other threads and join the caller to the current thread? I'd say it would make things like this a lot easier and * even more flexible ;) Thanks, Peter Spikings This message has been comprehensively scanned for viruses, please visit http://virus.e2e-filter.com/ for details. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WRT54GP2 SIP server on LAN port
Some more information if that might give anyone some ide what can be wrong.
WRTG54GP2 settings under Line1 that are set to something:
User ID: 100
Authentication Password: xxx
Registration / Proxy Server: 192.168.15.10
NAT Traversal: None
Under the router ip/Voice_adminPage.htm secret page and the tab Line
1 the following information might be of interest:
NAT Mapping Enable: No
SIP Port: 5060
EXT SIP Port: empty
SIP Proxy-Require: emtpy
Proxy: 192.168.15.10
Outbound Proxy: empty
Register: Yes
Use DNS SRV: No
Info tab:
Product Name: WRT54GP2-NA
Software Version: 3.1.3(LI)
Any of the above configuration that is wrong?
Or any setting that I have not typed out that are interesting?
I have tried using iptraf trying to see if there are any connection
attempts but there is nothing registred from the LInksys router.
So I'm currently stuck with no ideas what is wrong and how to fix it.
(se below for more details)
Anyone?
Thanks,
~Johannes
Thanks for the information Sherwood.
Then the question I had if the normal routing works for the SIP proxy
works with a LAN server.
But I cant get a success in connecting the router LINE1 to Asterisk.
WRT54GP2 says as status Can't connect to login server and there is no
connection attempt when running sip debug with verbose 4.
In my sip.conf this is specified:
[linksys]
type=friend
host=dynamic
username=100
secret=x
canreinvites=no
context=outgoing-sip
And in extensions.conf
[default]
exten = s,1,Dial(SIP/linksys|30|gr)
exten = s,2,VoiceMail(u100)
exten = s,3,Congestion
[outgoing-sip]
exten = _[0-9#*].,1,Dial(SIP/blixtvik-sip/${EXTEN}||t)
Now incoming calls gets the following loggs:
-- Executing Dial(SIP/0755xx-5499, SIP/linksys|30|gr) in new
stack
Sep 26 19:55:34 NOTICE[5525]: app_dial.c:777 dial_exec: Unable to create
channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing VoiceMail(SIP/0755xxx-5499, u100) in new stack
-- Playing 'vm-theperson' (language 'se')
-- Playing 'digits/1' (language 'se')
== Spawn extension (default, s, 2) exited non-zero on
'SIP/0755xxx-5499'
Sep 26 19:55:37 ERROR[5525]: cdr_sqlite.c:136 sqlite_log: cdr_sqlite:
attempt to write a readonly database
Sep 26 19:55:37 ERROR[5525]: cdr_csv.c:222 csv_log: Unable to re-open
master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied
The answering machine works but it will not get connected with my
WRT54GP2.
See anything that causes WRT54GP2 not to be able to register to Asterisk?
~Johannes
Actually, just point the line you want to use to a local ip address (the
asterisk server). I currently do this with my service. i.e. If your
Asterisk
server is 192.168.15.200, just make the proxy for line 1 that address.
It
routes internally just fine.
Sherwood McGowan
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Sunday, September 25, 2005 5:45 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] WRT54GP2 SIP server on LAN port
what I do is loopback the WAN port to a LAN port and am able to use both
(ie) take a cable from the wan port of the router and plug it into the
lan
port on the same router. This will give you a local ip and it still
should
allow connection out to your other provider.
On 9/25/05, Johannes [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
I'm trying to set up Asterisk behind my WRT54GP2 router that has a
intergrated ATA box.
My box are not locked in any way so I can access and change all
settings.
Now to the problem...
I have gotten Asterisk to register with my provider and everything works
just well..
Now it's time to get the intergrated ATA to connect to asterisk.
But the asterisk box in located on the LAN ports of the WRT54GP2.
I can't get the router to connect to Asterisk.
The question is then if the router does not use the normal routing table
and will force the connect to the SIP gateway to the WAN port even that
I
specified a LAN IP as the gateway.
Has anyone set up the WRT54GP2 to connect to a asterisk server thats on
the LAN ports with a LAN IP? Or is this impossible?
Regards,
~Johannes
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Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] I got 403, Forbidden... please help
Hi Harry, I tried your suggestion and it worked. But I don't hear any voice from the anonymous user. I don't hear the voice prompt? What should I do? Thanks, Ryan harry gaillac wrote: Hello, Try insecure=very in [sip.philonline.com] Harry --- Ryan Pagquil [EMAIL PROTECTED] a écrit : Hi, I'm setting up Asterisk as a voicemail with SER. My problem is, when a caller that is not registered with asterisk (no username and password in sip.conf) it prompts 403, Forbidden . I need all calls from outside of my network to reach asterisk for my users' voicemails, because anonymous users will surely reach voicemail of my users to leave messages. What do I need to do to make those anonymous callers to reach the voicemails of my users? here is my sip.conf. [general] port = 5060 bindaddr = 202.84.24.47 context = sip disallow=all allow=ulaw allow=alow ;register=me:[EMAIL PROTECTED]/1000 [sip.philonline.com] type=friend host=sip.philonline.com fromuser=rpagquil secret=test123 fromdomain=sip.philonline.com [phone1] type = friend username = phone1 secret = test123 host = dynamic context = sip mailbox = callerid=Test1 [acjeff] type=friend username=acjeff host=dynamic defaultip=10.0.1.236 nat=yes context=sip mailbox= callerid=Test2 [usser1] type = friend username = usser1 secret = test123 nat=yes host = dynamic context = sip mailbox = 111 callerid=User1 Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] radius and *
any one know where to get a radius module to work with the * sip server so SIP auth and Call accountingcan also bedone by radius? thanks! Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best drivers for HFC-S ISDN cards
I said: I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). Many thanks to those who replied. General consensus seems to be switching to mISDN or CAPI won't solve the intermittent echo problem. A follow-up with some more config information: Zaptel.conf span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,3,ccs,ami bchan=4-5 dchan=6 loadzone=uk defaultzone=uk Zapata.conf: [channels] language = en pridialplan = dynamic nationalprefix = 0 internationalprefix = 00 switchtype = euroisdn signalling=bri_cpe_ptmp group=1 context=isdn channel = 1-2 switchtype = euroisdn signalling=bri_cpe_ptmp group=2 context=isdn channel = 4-5 echocancel=yes echocancelwhenbridged=yes echotraining=no ;echotraining=800 rxgain=0.0 txgain=0.0 immediate=yes I've tried echotraining off, on, 100, 400 and 800, none of which seem to help matters very much. Any suggestions for getting rid of echo on these lines would be gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited Tel: (07010) 710715 Mobile: (07811) 332969 Skype: minotaur-uk ICQ: 13350579 AIM: MinotaurUK MSN: [EMAIL PROTECTED] Y!: Minotaur_Chris This email is made from 100% recycled electrons ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Turn off echo-cancellation when fax is detected?
How can I do this? Ive set faxdetect=both in zapata.conf. Does this cancel echo-cancellation (echo-training) when a fax is detected or is this just for using exten=fax, in extensions.conf.? Im having trouble getting spanDSP - RxFax to recieve faxes. I am using Asterisk 1.0.8 and the fax number is registered in sip.conf Thanks, Arne Morten ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * Accounting with Oracle
Hello all, I use the asterisk with a oracle db in th ebackend. I want to use the db for accounting also. I saw that AMP has a mysql table with the accounting datas. Isit possible to por this to oracle or does anybody has a accounting agi or whatever which uses oracle? Regards Rene ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Best drivers for HFC-S ISDN cards
Mine is very similar: i don't have echocancelwhenbridged=yes because it seems work only on TDM, is it ? And in Italy, I often have set pridialplan = unknown About echo I have some problems, but only at the beginning of the call. After 3-4 seconds the echo became almost null, specially with snom 190; with pa168s and ywh10 I have again some problem, the echo come up also after 1 minute of conversation. The most strange think is that on older version of bristuff, with same configuration files, I never had this problem. Any suggestion? Specially for echo problem ? Thanks all Giordano Grandis [EMAIL PROTECTED] -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Chris Bagnall Inviato: martedì 27 settembre 2005 12.07 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: [Asterisk-Users] Best drivers for HFC-S ISDN cards I said: I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). Many thanks to those who replied. General consensus seems to be switching to mISDN or CAPI won't solve the intermittent echo problem. A follow-up with some more config information: Zaptel.conf span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,3,ccs,ami bchan=4-5 dchan=6 loadzone=uk defaultzone=uk Zapata.conf: [channels] language = en pridialplan = dynamic nationalprefix = 0 internationalprefix = 00 switchtype = euroisdn signalling=bri_cpe_ptmp group=1 context=isdn channel = 1-2 switchtype = euroisdn signalling=bri_cpe_ptmp group=2 context=isdn channel = 4-5 echocancel=yes echocancelwhenbridged=yes echotraining=no ;echotraining=800 rxgain=0.0 txgain=0.0 immediate=yes I've tried echotraining off, on, 100, 400 and 800, none of which seem to help matters very much. Any suggestions for getting rid of echo on these lines would be gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited Tel: (07010) 710715 Mobile: (07811) 332969 Skype: minotaur-uk ICQ: 13350579 AIM: MinotaurUK MSN: [EMAIL PROTECTED] Y!: Minotaur_Chris This email is made from 100% recycled electrons ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax
Does anyone have any experience with Teliax for inbound IAX? Been working fine for me for over six months with multiple did's over iax. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma and Digium same machine?
I'm using a Sangoma A101 card alongside an older TDM400 and they seem to be playing nice. I've had it in production for a few months now with no problems. Thanks, Reid Forrest, CISSP Max-IS Inc. [EMAIL PROTECTED] Direct/Cell: 321-214- Main: 407-786-9600 -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of William Lloyd Sent: Monday, September 26, 2005 1:08 PM To: asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sangoma and Digium same machine? Anybody ever put a Sangoma and a Digium card in the same server? Specifically a four port card from each company? -bill [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Best drivers for HFC-S ISDN cards
Well done Tim...could u post here your Zapata.conf ? :) I'm in Italy and have some issues with echo Thanks Giordano Grandis [EMAIL PROTECTED] -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Tim Robinson Inviato: lunedì 26 settembre 2005 22.30 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Best drivers for HFC-S ISDN cards Chris I have only ever used zaphfc drivers and for me they are perfect. Echo has never been a problem. It would be helpful if you were to provide a bit more information to the group about your configuration so we can try and help you work out the cause. Switching to capi or mISDN is unlikely to help and will almost certainly be a retrograde step as far as I hear from these forums. Best regards Tim Robinson Basingstoke, UK Chris Bagnall wrote: It seems that HFC-S cards can be connected with asterisk in a few different ways - isdn4linux, mISDN, chan_capi or zaphfc. I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). What's the recommended way to hook up these ISDN cards? Is switching to capi or mISDN likely to remove the echo problem completely, or is this one of those things one has to accept? Thanks in advance. Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax problem
On Mon, Sep 26, 2005 at 11:02:47AM -0600, Rich Adamson wrote: For #2, incoming calls would be handled with: exten = 6789,1,Dial(SIP/1235) Besides that : *CLI iax2 show registry Host UsernamePerceived Refresh State X.X.X.X:4569 Username1 [MYIP]:456960 Registered X.X.X.X:4569 Username2 [MYIP]:456960 Registered X.X.X.X:4569 Username3 [MYIP]:456960 Registered source and destination ports for all 3 iax registrations are the same , and my isp see only one, becouse rest is overwriten. Have you tried using three different contexts for those in iax.conf? Yes and result is as I suppose : -- Accepting UNAUTHENTICATED call from X.X.X.X: requested format = ilbc, requested prefs = (ilbc|gsm|ulaw|alaw), actual format = ilbc, host prefs = (), priority = caller -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1, IAX2/1237) in new stack -- Called 1237 -- Call accepted by 192.168.57.238 (format gsm) -- Format for call is gsm -- IAX2/1237-8 is ringing -- Hungup 'IAX2/1237-8' Everything enters via last registred username 'Username3'. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wait before accepting the call
hello! i'm looking for a way to prolonge a pstn-call for 5 seconds before it enters the extensions.conf. this is for testing purposes, all numbers of a ddi should be received by asterisk before the call is walking through the extensions. how can i achive this? i've not seen a feature like this for zapata or zaptel, does anyone have an idea how this could be done? thx christian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Integration with NMS AG-E1/T1
What is the line protocol you're using on this legacy PBX? Is it EM Wink? If so, then you'd just configure the Digium card for wink, plug in a T1 crossover cable and you should be ready to start testing. On 9/27/05, Exciting [EMAIL PROTECTED] wrote: I want to replace a custom PBX, that is infront on a IVR system based on OLD NMS AG-E1 Card.The Cards is configurated with CAS Digitalmode, someone can give me some info about Digim Cards CAS configurationi need a conversion Table? I wanto to don't touch configuration on winbox, i want only replace HWPBX box with asterisk.DiagramTelco E1 ===Proprietary PBX(CAS)===IVR Server AG-E1RegardsConfigurations ag.cfgIdleCode = 0xD5, 0x5DigitalMode= CASClockRef = NET1_Get free infected, boring, wrong, empty, or any other email for yourself. Go to --- http://www.mailchoose.com___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax
Jason Schafer wrote: Does anyone have any experience with Teliax for inbound IAX? Yes, have many accounts. Very good service and support. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] MS Live Communications Server
I still find out how to let LCS 2005 accept SIP invite from Asterisk, Need more help. Hi jacky, can you please share your experience and explain how to let LCS accept SIP invite from Asterisk. I deseperate trying to place a call from asterisk to LCS. (calling from Asterisk to LCS using TCP_SUPPORT seems to work fine) thx in advance 2005/8/13, bubuk [EMAIL PROTECTED]: Hi, I already posted this in the user list, but this list is probably the better one. My question was: Does anyone played around with the LCS and Asterisk? Because the LCS is doing no RFC compliant SIP, i wonder if it can work. Google couldn't tell me. If someon heard about that, please let me know. Thank you Volker ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Dialtone problems with phpagi and asterisk
Does nobody know a solution or an approach to a solution? Michael Michael Häberle wrote: Hi there In our php-application we use phpagi to communicate with asterisk (as the voip-client we use x-pro) Sometimes it occurs that the dialtone is very choppy or not present. If we dial directly in x-pro this problem has never occured. I dont know what the problem is, first I thought it is the bandwith (which is actually a problem), but if that would be the major problem it wouldnt work in x-pro either, I assume. Another problem is that sometimes after two or three times ringing the phone hangs up. No idea what the problem is. (this problem does not occur with x-pro directly) We use phpagi 2.14 Suse Linux 8.x I dont know the asterix version (we downloaded it in july 2005) Michael -- Immosky AG Service-Zentrale Dufourstr. 5 CH-8702 Zollikon-Zürich Tel+41 (0)43 344 52 52 Fax +41 (0)43 344 52 58 www.immosky.ch [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Integration with NMS AG-E1/T1
I don't kwnow if between NMS and PBX there's a EM Wink protocol, do you have some info to retreive it from ag.cfg (NMS board config file) ? The PBX is connected to an E1,(Italy) (ITU-T G.703 E1), and also to 2 AG-E1 machines, the pbx route incoming calls to an proprietary ivr on these machines. The are some difference into two ag.cfg files on each IVR. Do you have some suggestion about discover line protocol between pbxivr in any way? Do you have confience with AG-NMS Card ? I appreciate your interest. Regards --- BJ Weschke [EMAIL PROTECTED] wrote: From: BJ Weschke [EMAIL PROTECTED] Date: Tue, 27 Sep 2005 08:03:43 -0400 To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Integration with NMS AG-E1/T1 What is the line protocol you're using on this legacy PBX? Is it EM Wink? If so, then you'd just configure the Digium card for wink, plug in a T1 crossover cable and you should be ready to start testing. On 9/27/05, Exciting [EMAIL PROTECTED] wrote: I want to replace a custom PBX, that is infront on a IVR system based on OLD NMS AG-E1 Card. The Cards is configurated with CAS Digitalmode, someone can give me some info about Digim Cards CAS configuration i need a conversion Table? I wanto to don't touch configuration on winbox, i want only replace HWPBX box with asterisk. Diagram Telco E1 ===Proprietary PBX(CAS)===IVR Server AG-E1 Regards Configurations ag.cfg IdleCode = 0xD5, 0x5 DigitalMode= CAS ClockRef = NET1 _ Get free infected, boring, wrong, empty, or any other email for yourself. Go to --- http://www.mailchoose.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get free infected, boring, wrong, empty, or any other email for yourself. Go to --- http://www.mailchoose.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 hard phone
I purchased an IAX2 hardphone, X100 otherwise known as a Netweb X100 or YWH100 with a PA168 chip and the latest firmware 1.45 available, from a US retailer. I was able to configure the phone to work with my Asterisk box, except the hold and transfer buttons do not work. When you press the hold button, it rings endlessly, the transfer button, displays transferring but it does nothing. Has anybody with these phones run into similar problems? Or can recommend a good functional IAX2 hard phone. Thanks, Alberto The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Integration with NMS AG-E1/T1
What do your NetworkInterface.T1E1[X..X] and TCPFiles[X] lines in ag.cfg look like? Yes. I've worked with the AG series boards from NMS before. On 9/27/05, Exciting [EMAIL PROTECTED] wrote: I don't kwnow if between NMS and PBX there's a EM Wink protocol, do you have some info to retreive it from ag.cfg (NMS board config file) ?The PBX is connected to an E1,(Italy) (ITU-T G.703 E1), and also to 2 AG-E1 machines, the pbx route incoming calls to an proprietary ivr on these machines.The are some difference into two ag.cfg files on each IVR.Do you have some suggestion about discover line protocol between pbxivr in any way?Do you have confience with AG-NMS Card ?I appreciate your interest.Regards --- BJ Weschke [EMAIL PROTECTED] wrote:From: BJ Weschke [EMAIL PROTECTED]Date: Tue, 27 Sep 2005 08:03:43 -0400 To:[EMAIL PROTECTED],AsteriskUsersMailingList-Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Integration with NMS AG-E1/T1Whatis the line protocol you're using on this legacy PBX? IsitEMWink? If so, then you'd just configure the Digium cardforwink, plug in a T1 crossover cable and you should be ready to start testing.On 9/27/05, Exciting [EMAIL PROTECTED] wrote:Iwanttoreplacea custom PBX, that is infront on a IVRsystem based on OLD NMS AG-E1 Card. TheCards is configurated with CAS Digitalmode, someone cangivemesomeinfoabout Digim Cards CAS configurationineed a conversion Table?Iwanto to don't touch configuration on winbox, i want only replace HWPBX box with asterisk.DiagramTelco E1===ProprietaryPBX(CAS)===IVRServerAG-E1RegardsConfigurations ag.cfgIdleCode = 0xD5, 0x5DigitalMode= CAS ClockRef = NET1_Getfree infected, boring, wrong, empty, or any other emailfor yourself. Go to --- http://www.mailchoose.com___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users_Get free infected, boring, wrong, empty, or any other email for yourself. Go to --- http://www.mailchoose.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax
Yes, and I posted the information on the Wiki. Regards, Chris - Original Message - From: Chris Mason (Lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 27, 2005 7:15 AM Subject: Re: [Asterisk-Users] Teliax Jason Schafer wrote: Does anyone have any experience with Teliax for inbound IAX? Yes, have many accounts. Very good service and support. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to create channel of type 'Zap'
I installed Asterisk 1.0 CVS on a Debian Sarge System. I am using two ISDN-HFC-Cards and a point-to-point ISDN Connection. Everything seemed to work pefectly. But today I realized that I cannot use two lines at the same time. I get the error message: 3 active channel(s) asterisk*CLI show channels Channel (ContextExtensionPri ) State Appl. Data Zap/2-1 (internS0 555858 2 )Ring (None) (None) SIP/user1-e4eb (from-sip1 ) Up Bridged Call Zap/4-1 Zap/4-1 (externS0 4445858 1 ) Up Dial SIP/user1|20|t Sep 27 15:13:56 NOTICE[13491]: app_dial.c:805 dial_exec: Unable to create channel of type 'Zap' I restarted Asterisk and it worked for 1 or 2 times. After that I had the same problem. Any hints, where I can start searching? Is there any possiblity to force the channel to hangup or something like that? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskJava - Queue
Thanks for your response. AFAIK I can redirect, bridge, drop and answer a call but I can't find the way to do, for example: - Get the call back from the queue, play a message and put it again in the queue. and - Get a linked call (caller to Agent), unlink it (releasing the agent) and play something to the caller. thanks in advance Sebas Alexander Lopez wrote: You may loose 'control' of the call but you can always 'get it back' Use the UnigueID of the call to track it throught Asterisk. You can palce a monitor event to redirect, bridge, drop, answer or antything else. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Silva Sent: Monday, September 26, 2005 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] AsteriskJava - Queue Hi, I am using AsteriskJava and I have some problems, I will appreciate any help... My system has the following architecture (in the server side): - An app server (connected to the asterisk console) - An AGI Server (developed with AsteriskJava) - An AGI Script (executed by the above AGI Server) In the client side (Agents answering call center calls): - A softphone - A client program (used to search and register call details) Here is the thing: - From AGI Server I detect that a call is coming from PSTN and launch the AGI Script - From AGI Script I put the call in the queue and I loose the control of the call (here is my first confusion) - The agent answer the call (using his/her softphone) and I get the event from the Asterisk Console with my App Server. Now, I need to play something (TTS, wav, etc) to the caller based on the client application wich is connected to my App Server. What I want you to know is that the information to be played to the caller comes from an external source. So, my two big questions/confusions are: - How can I get the entire control of the call depending on the status of the call, for example, if the call is in the queue and I need to play or do something with it, where and how I have the control? until now, when I put the call in the queue I loss the control until the caller or the agent hangs the call. - Once the call is answered by the Agent, how can I unlink the two channels (releasing the agent) to let the caller hear the text that the agent sent. Thanks in advance, Sebas -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Moaning dog...
Here's one for you phone people An elderly lady phoned her telephone company to report that her telephone failed to ring when her friends called - and that on the few occasions when it did ring, her pet dog always moaned right before the phone rang. The telephone repairman proceeded to the scene, curious to see this psychic dog or senile elderly lady. He climbed a nearby telephone pole, hooked in his test set, and dialed the subscriber's house. The phone didn't ring right away, but then the dog moaned loudly and the telephone began to ring. Climbing down from the pole, the telephone repairman found: 1. The dog was tied to the telephone system's ground wire via a steel chain and collar. 2. The wire connection to the ground rod was loose. 3. The dog was receiving voltage of signaling current when the phone number was called. 4. After a couple of such jolts, the dog would start moaning and then urinate on himself and the ground. The wet ground would complete the circuit, thus causing the phone to ring. Which demonstrates that some problems CAN be fixed by pissing and moaning. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Dialtone problems with phpagi and asterisk
Without information about your dialplan and what the phpagi script does there is not much anyone can do. I do not know of any known issues that may account for the problem you are having. Update with further information and maybe someone will be able to provide some insight. --johann Michael Häberle wrote: Does nobody know a solution or an approach to a solution? Michael Michael Häberle wrote: Hi there In our php-application we use phpagi to communicate with asterisk (as the voip-client we use x-pro) Sometimes it occurs that the dialtone is very choppy or not present. If we dial directly in x-pro this problem has never occured. I dont know what the problem is, first I thought it is the bandwith (which is actually a problem), but if that would be the major problem it wouldnt work in x-pro either, I assume. Another problem is that sometimes after two or three times ringing the phone hangs up. No idea what the problem is. (this problem does not occur with x-pro directly) We use phpagi 2.14 Suse Linux 8.x I dont know the asterix version (we downloaded it in july 2005) Michael ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IBM x306 - some progress
I dug up my Netfinity ServeRaid readme: Power on your system and observe the screen. Press F1 when the Press F1 for Configuration/Setup and Press F2 for Diagnostics messages appear. The Configuration/Setup Utility main menu will appear. Select Advanced Setup using the Up or Down arrow key and press Enter. Select System Service Processor Settings using the Up or Down arrow key and press Enter. --Change System Service Processor Hardware Interrupt from Autoconfigure to IRQ 5. -- Press Esc. Select PCI Bus Control using the Up or Down arrow key and press Enter. Select Planar Device PCI Interrupt Routing using the up or down arrow keys and press Enter. --Change Planar Raid IRQ from Autoconfigure to an available IRQ using the Left or Right arrow key.-- Notes: If IRQ 10 is available, use IRQ 10. If a PCI RAID adapter card is also installed on your system, select Slot Device PCI Interrupt Routing using the Up or Down arrow key and press Enter. Change the IRQ for the slot used from Autoconfigure to an available IRQ (you can share the Planar RAID IRQ). In my Netfinity w/ TDM400, I have the TDM set to IRQ 15, the IDE controller disabled (don't care about the CD-ROM), my PRI TDM card to IRQ 11, VGA set to IRQ3 (COM ports disabled) and my ServeRaid set to IRQ9. IRQ 5 is used for the System Service Processor. IIRC all this I set in the F1 Setup. Rule of thumb best practice is to disable any hardware in the system that isn't needed specifically: USB, COM ports, parallel and this should give you plenty of elbow room to juggle interrupts. hth -Original Message- From: Nir Simionovich [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 27, 2005 1:48 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IBM x306 - some progress Hi Marco, As far as I can recall, the IBM setup utility can enable you to change the IRQ of the SCSI controller. In addition, I've never seen a WildCard board bound to IRQ7 on any box, which is very weird in it self. I'm flying over to Ireland today (actually, at the airport right now), and I'm coming back on Sunday. If You'd like, you can bring your box to my office after Rosh-Hashana, and I'll try to help you out. Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Supino Sent: Monday, September 26, 2005 10:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IBM x306 - some progress Hi, I asked yesterday about a problem with x306 and IRQ sharing, didnt get much info, now, i was playing with lspci, and see something strange, lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is also on IRQ 7, lspci -bv (from the man - b - shows bus-centric view, as seen by the BUS and not by the kernel) shows me the TDM400P is on IRQ 5, why does the kernel puts it on IRQ 7 ? any insights much appriciated. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax problem
For #2, incoming calls would be handled with: exten = 6789,1,Dial(SIP/1235) Besides that : *CLI iax2 show registry Host UsernamePerceived Refresh State X.X.X.X:4569 Username1 [MYIP]:456960 Registered X.X.X.X:4569 Username2 [MYIP]:456960 Registered X.X.X.X:4569 Username3 [MYIP]:456960 Registered source and destination ports for all 3 iax registrations are the same , and my isp see only one, becouse rest is overwriten. Have you tried using three different contexts for those in iax.conf? Yes and result is as I suppose : -- Accepting UNAUTHENTICATED call from X.X.X.X: requested format = ilbc, requested prefs = (ilbc|gsm|ulaw|alaw), actual format = ilbc, host prefs = (), priority = caller -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1, IAX2/1237) in new stack -- Called 1237 -- Call accepted by 192.168.57.238 (format gsm) -- Format for call is gsm -- IAX2/1237-8 is ringing -- Hungup 'IAX2/1237-8' Everything enters via last registred username 'Username3'. I'm out of ideas other then to open a feature request to add the /1234 syntax to the register statement for iax. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 hard phone
Alberto, PA168 chip does not have Hold and Transfer features on it until firmware version 1.44. Atcom never claimed that these will work as the Pa168 firmware is still under development. Yesterday I met Peter Sun, President and owner of Atcom China, in New York. He is here toattend VON in Boston. I have enquired about the fix for this and he said that Hold and Transfer are working with 1.45 Firmware. I mentined that this is not the case with the phones we tried to use here. Peter mentioned that firmware version 1.46 is going to be relased this week, which will provide these features and also the Voice Mail Messages Indicator Led should work too. I am waiting for this firmware to be released. I will forward this to you as soon as it is released. In the meantime you can check for the updates at http://www.iareaphone.com under downloads, if this becomes available sooner. Seshu Kanuri From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto RiscoSent: Tuesday, September 27, 2005 9:06 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] IAX2 hard phone I purchased an IAX2 hardphone, X100 otherwise known as a Netweb X100 or YWH100 with a PA168 chip and the latest firmware 1.45 available, from a US retailer. I was able to configure the phone to work with my Asterisk box, except the hold and transfer buttons do not work. When you press the hold button, it rings endlessly, the transfer button, displays transferring but it does nothing. Has anybody with these phones run into similar problems? Or can recommend a good functional IAX2 hard phone. Thanks, Alberto The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pbx_wilcalu.so: undefined symbol:
Pikoro wrote: Anyone run into this? This is from the latest 1.2.0 beta1 tarball. Got it all compiled, but this undefined symbol is stopping asterisk from loading. When you change major versions, before you install you should: rm -rf /usr/lib/asterisk/modules/* I also rm -rf /usr/include/asterisk/* Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Early Media in 180 Ringing
Ronald Voermans wrote: If guess I figured it out already. I made some changes in chan_sip.c (when ringing was received, it didn't check for SDP), and recompiled. I don't know what all of this means, but I'm sure it could be of value to others. Can you submit your patch to bugs.digium.com? Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] function LEN missing
I'm running asterisk 1.2b1 and all seems to be workingright in general. I load modules explicitly in modules.conf, and since my upgrade ast 1.09 I have only one problem: The LEN function (length of string). What module do I need to load to get this string handling function? Thanks MD ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 405 Method Not Allowed error
Hi everybody, I'm curious to know what this message generally indicates. I have XLite softphones on two different machines accessing the Asterisk server behind a NAT. The server is able to find them just fine, but instead of registering when Asterisk reloads they return this message back to Asterisk. I feel like I've looked everywhere but there doesn't seem to be an explanation... Thanks, Tim __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Software only Asterisk PBX (commercial)
Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: channel offhook state
We had the same thing until we started using Voicetronix, it seems that this happens when calls collide i.e... incoming call with an outgoing? We added a script that did a soft hang-up after a call was ended and that seemed to work ok. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, September 26, 2005 6:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FW: channel offhook state Has anyone else experienced the same problem, where a Zap channel gets stuck in off-hook state? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 1:45 PM To: asterisk-users@lists.digium.com Subject:[Asterisk-Users] FW: channel offhook state -Original Message- From: Jacqueline Lee [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 11:46 AM To: asterisk-users@lists.digium.com Subject:channel offhook state We are using a digium card (TDM400) with asterisk for our access to the PSTN. Initially when the server starts, all the zap channels on the card are in the onhook state. As soon as a channel is used (for inbound or outbound PSTN calls) the corresponding channel goes into offhook state, and stays in offhook state, even after the call ends; Asterisk log shows that the channel was hungup. Most of the time, the channel is still usable to make more PSTN calls, even though it shows in offhook state. Occasionally the channel becomes unusable for making PSTN calls (usually channel 1). The symptom is Asterisk and the client show the PSTN call was established, but the destination PSTN number never really receives the call. Shouldn't the channel go back to onhook state once the call hangs up? Is the persistent offhook state causing the channel to eventually become unusable? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.5/110 - Release Date: 9/22/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.5/110 - Release Date: 9/22/2005 File: ATT00068.txt -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/2005 File: ATT00158.txt attachment: winmail.dat___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk European Digital CAS Help
Someone can give me more info about Asterisk European Digital CAS , I need to make talk asterisk with a AG-E1 card with this protocol. (TCP=euc0.tcp); Is built in supported or i need some patch ? Regards _ Get free infected, boring, wrong, empty, or any other email for yourself. Go to --- http://www.mailchoose.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Dialtone problems with phpagi and asterisk
Ok :)
the dialplan looks like that (mynumber is a tel-number):
-
[general]
static=yes
writeprotect=no
[telout]
exten = _X.,hint,SIP/41
exten = _X.,1,dial(SIP/${EXTEN})
exten = _X.,2,SetCIDName(anonymous)
exten = _X.,3,dial(SIP/[EMAIL PROTECTED],30,r)
exten = _X.,4,Hangup
-
I dial out of a webapplication, when I press a button, we connect to
asterisk through phpagi.
here are the php-functions:
function startCall($number,$uid) {
$returnValue = false;
$state = getStatus();
if ($state = 0 $state 4) {
$asm = new AGI_AsteriskManager();
if($asm-connect())
{
$call = initCall($asm, $number);
$asm-disconnect();
if (trim($call['Response']) == Error) {
$returnValue = false;
} else {
$returnValue = true;
}
} else {
echo Connect to Asterisk FAILED;
}
} else {
echo Extension already in use;
}
function initCall($asm, $number) {
$call = $asm-send_request('Originate',
array('Channel'=SIP/ . $_COOKIE['extension'],
'Context'='telout',
'Exten'=$number,
'Priority'=1,
'Timeout'=3,
'Async'=false,
'Callerid'='anonymous'));
return $call;
}
for the cookie we have defined a channel in sip.conf.
Later we start to monitor the call (writing *.wav files)
Dont know if that causes the described problems.
If the connection is made an the user on the other side of the line
takes the phone, we phone with x-pro.
Johann wrote:
Without information about your dialplan and what the phpagi script does
there is not much anyone can do. I do not know of any known issues that
may account for the problem you are having.
Update with further information and maybe someone will be able to
provide some insight.
--johann
Michael Häberle wrote:
Does nobody know a solution or an approach to a solution?
Michael
Michael Häberle wrote:
Hi there
In our php-application we use phpagi to communicate with asterisk (as
the voip-client we use x-pro)
Sometimes it occurs that the dialtone is very choppy or not present.
If we dial directly in x-pro this problem has never occured.
I dont know what the problem is, first I thought it is the bandwith
(which is actually a problem), but if that would be the major problem
it wouldnt work in x-pro either, I assume.
Another problem is that sometimes after two or three times ringing
the phone hangs up. No idea what the problem is. (this problem does
not occur with x-pro directly)
We use phpagi 2.14
Suse Linux 8.x
I dont know the asterix version (we downloaded it in july 2005)
Michael
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Tel+41 (0)43 344 52 52
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RE: [Asterisk-Users] Software only Asterisk PBX (commercial)
Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Have a look at the AMP project http://sourceforge.net/projects/amportal ~ron ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] IAX2 hard phone
Hello Alberto, You must upgrade the firmware by taking the last one at www.aredfox.com which is the PA168 manufacturer. Mine Ip-phones are running well with IAX2 and flash hook for transferts. Good luck. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Alberto Risco Envoyé : mardi 27 septembre 2005 15:06 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] IAX2 hard phone I purchased an IAX2 hardphone, X100 otherwise known as a Netweb X100 or YWH100 with a PA168 chip and the latest firmware 1.45 available, from a US retailer. I was able to configure the phone to work with my Asterisk box, except the hold and transfer buttons do not work. When you press the hold button, it rings endlessly, the transfer button, displays transferring but it does nothing. Has anybody with these phones run into similar problems? Or can recommend a good functional IAX2 hard phone. Thanks, Alberto The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP Buster stopped working?
Hi, I was successfully using VoIP Buster via IAX2 for several weeks now. Yesterday/today it spontaneously stopped working. Using the real client the connection works well though. Anybody else experiencing this problem? Or asked differently: Is there anybody for whom it is still working? Can anybody tell me what the problem could be from this: -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, IAX2/[EMAIL PROTECTED]/0049712147557) in new stack -- Called [EMAIL PROTECTED]/0049712147557 -- Call accepted by 213.61.187.156 (format alaw) -- Format for call is gsm -takes a long while ~15 to 30 sec here-- -- Hungup 'IAX2/voipbuster/3' == No one is available to answer at this time Cheers, Arik ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to change ${VM_DATE} in voicemail.conf
Hi all,
I don't find where you can setup the date (${VM_DATE}) in french for the mail.
Is anybody can help me?
Amaury BOSSÉ
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Re: [Asterisk-Users] Extension availabilty
http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20extensions http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20BristuffDevstate Can anyone say for certain what asterisk version introduced the hint priority? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] analogue phone with asterisk
I am a newbee to asterisk. I recently installed [EMAIL PROTECTED]. Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a PSTN phone connection. I want to use my analogue phones as the end points for my asterisk box to make and receive calls. All i want is to use my analogue phones instead of soft phones. Can some one help me what hardware interface i need for that and how should i go about it? if there is any HOW-TO for that it will be of great help. thanks, rajesh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] MS Live Communications Server
Hi, Richard, I still try, but fail with rtp transfer. 2005/9/27, richard Coco [EMAIL PROTECTED]: I still find out how to let LCS 2005 accept SIP invite from Asterisk, Need more help. Hi jacky, can you please share your experience and explain how to let LCS accept SIP invite from Asterisk. I deseperate trying to place a call from asterisk to LCS. (calling from Asterisk to LCS using TCP_SUPPORT seems to work fine) thx in advance 2005/8/13, bubuk [EMAIL PROTECTED]: Hi, I already posted this in the user list, but this list is probably the better one. My question was: Does anyone played around with the LCS and Asterisk? Because the LCS is doing no RFC compliant SIP, i wonder if it can work. Google couldn't tell me. If someon heard about that, please let me know. Thank you Volker ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] failed make install on Solaris 10
I am sure you might have tried adding the current directory to the PATH variable. I never compiled asterisk on solaris, but it seems to be working for my other applications. regards, rajesh - Original Message - From: Joseph Rothstein [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, September 27, 2005 4:19 AM Subject: [Asterisk-Users] failed make install on Solaris 10 I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found anywhere! make: *** [bininstall] Error 2 [EMAIL PROTECTED] # I would prefer not to install everything manually if possible. Anyone have any ideas how I get around this? Asterisk is clearly in the directory, but for some reason Solaris can't pick it up. Regards, Joe --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Sep 26, 2005 at 10:00:10PM -0400, Patrick wrote: I know it seems basic but did you make sure and plug power into the board when you installed it into the PCI slot? I spent about three hours trying to get the dang thing to work in my machine until I decided to stick the card into another PCI slot. That is when I noticed that I had forgotten to ALSO plug power into the board from the power supply. Everything worked fine after that (yep, I was a noob). :-) You get a messge about it from the module at module load time. rmmod wctdm (or wcfxo) and re- modprobe it, and then run: dmesg |tail -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -- Message: 19 Date: Tue, 27 Sep 2005 00:15:14 -0700 (PDT) From: somesh s [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Problem setting up TDM22B card To: Tzafrir Cohen [EMAIL PROTECTED] Cc: Asterisk Users asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi, I know some bits about zaptel.conf and zapata.conf but problem is modules wctdm, wcfxo, wcfxs are not getting loaded. modprobe zaptel is successful! Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 25, 2005 at 11:50:06PM -0700, somesh s wrote: Hi, I got the following output when I run the command. genzaptelconf -svd ./genzaptelconf: line 616: /etc/init.d/asterisk: No The script assumes that there is an /etc/init.d/asterisk script. Please stop asterisk manually. such file or directory Unloading zaptel modules: wcusb zaptel Test Loading modules: - zaptel - zaphfc - qozap Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wctdm Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcfxo Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcfxs - pciradio Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - tor2 Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - torisa Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wct1xxp Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the
[Asterisk-Users] [MSG]TDM Error on ASUS Pundit-R
Hi I have looked around but I cant find an answer for this, I randomly get the error 'TDM PCI Master abort' and the system locks up. All I have found so far are a couple other posts on it but no solution. Running fedora core 3, asterisk stable, zaptel stable. Any help will be appreciated. Morgan Gilroy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Problem setting up TDM22B card
[EMAIL PROTECTED] wrote on 09/27/2005 03:13:21 AM: Hi, I did dmesg | tail it says ... dmesg | tail f6 != 58 f7 != 59 f8 != 58 f9 != 59 fa != 58 fb != 59 fc != 58 fd != 59 fe != 58 Freshmaker failed register test The only time I've seen this it has been on a PCI 2.1 computer. On a PCI 2.2 computer, I did not see this. It also was a early TDM board. If you have a pre-Rev F board, you may want to swap it for a newer one. I am pretty sure that this error was fixed by moving from an earlier board to a Rev F. I have a Rev H now, with no issues. I have not been following this thread closely. Which chipset does your motherboard use? For the record, none of the desktop or server Intel 440-series support PCI 2.2. (Technically, a single mobile chipset, the 440MX, does support PCI 2.2) However, all of the 800-series chipsets do. The easy way to figure this out for Intel chipsets is: 1) Does the motherboard use slot processors? If so, it's PCI 2.1. 2) Does the motherboard support 133MHz PIII processors? If so, you're possibly PCI 2.2. 3) Pentium 4 chipsets are all PCI 2.2. I have no idea what other non-Intel chipsets support PCI 2.2. Reference: http://www.intel.com/design/chipsets/mature/450_440.htm http://www.intel.com/design/chipsets/mature/index.htm Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Software only Asterisk PBX (commercial)
Also check out http://www.bicom.us pretty expensive but if that's your thing :) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ronald Hartmann Sent: 27 September 2005 16:47 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Software only Asterisk PBX (commercial) Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Have a look at the AMP project http://sourceforge.net/projects/amportal ~ron ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 500 - problem dialing extra numbers
hi thereI'm setting up [EMAIL PROTECTED] and I'm using Polycom IP 500 phones.When I call a number that has a digital receptionist (i.e. "dial 1 or such and such, dial 2 for this and that...") the Polycom doesn't seem to send the extra digits. When I try it with X-Lite things appear to work fine, so I think the problem is with the Polycom configuration.Here's some of my configuration files. If I didn't included an important one please let me know.-.cfg-?xml version="1.0" standalone="yes"?!-- Default Master SIP Configuration File--!-- Edit and rename this file to Ethernet-address.cfg for each phone.--!-- $Revision: 1.13 $ $Date: 2004/11/26 23:30:44 $ --APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone1.cfg, sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="/log/" /-sip.cfg-?xml version="1.0" standalone="yes"?!-- SIP Application Configuration File --!-- $Revision: 1.63 $ $Date: 2004/11/08 18:52:16 $ --sip voIpProt local voIpProt.local.port=""/ server voIpProt.server.1.address="10.0.20.0" voIpProt.server.1.port="5060" voIpProt.server.1.transport="DNSnaptr" voIpProt.server.1.expires="300" voIpProt.server.1.register="1" voIpProt.server.1.retryTimeOut="0" voIpProt.server.1.retryMaxCount="0" voIpProt.server.1.expires.lineSeize="30" / SIP voIpProt.SIP.useRFC2543hold="1" voIpProt.SIP.lcs="0" voIpProt.SIP.sendCompactHdrs="0" voIpProt.SIP.WM50="0" voIpProt.SIP.keepalive.sessionTimers="0" voIpProt.SIP.requestURI.E164.addGlobalPrefix="" outboundProxy voIpProt.SIP.outboundProxy.address="10.0.20.0" voIpProt.SIP.outboundProxy.port="5060" / alertInfo voIpProt.SIP.alertInfo.1.value="" voIpProt.SIP.alertInfo.1.class="" / requestValidation voIpProt.SIP.requestValidation.1.request="" voIpProt.SIP.requestValidation.1.method="" voIpProt.SIP.requestValidation.1.request.1.event="" digest voIpProt.SIP.requestValidation.digest.realm="PolycomSPIP" / /requestValidation specialEvent voIpProt.SIP.specialEvent.lineSeize.nonStandard="1" voIpProt.SIP.specialEvent.checkSync.alwaysReboot="0"/ conference voIpProt.SIP.conference.address="" / /SIP /voIpProt dialplan dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1" digitmap dialplan.digitmap="" dialplan.digitmap.timeOut="3"/ routing server dialplan.routing.server.1.address="" dialplan.routing.server.1.port="5060"/ emergency dialplan.routing.emergency.1.value="911" dialplan.routing.emergency.1.server.1="1"/ /routing /dialplan logging level change log.level.change.sip="4" log.level.change.sip.obs="5"/ /level /logging/sipI just realized something... I don't have a phone1.cfg file, should I?I adopted this system in a partial working state from someone else and I'm still figuring out why things are the way they are.thanks-jachin___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One-way audio with VPN
I've got a one-way audio problem, but I've looked through a few documents on the subject and I'm not sure that it's the same issue. User A calls a local Asterisk user B via a public SIP gateway (voiptalk.org) using (sip:[EMAIL PROTECTED]) B is connected to the Asterisk server via VPN B is registered (and has successful bi-directional conversations with other users on the VPN) Asterisk correctly forwards the call via SIP and B's phone rings and is answered, but B can't hear A So there appears to be an audio-path blockage from A via Asterisk to B. Now if A leaves a voicemail message on the asterisk box, that's fine (the sound file contains a recording of A's voice!) Therefore, it looks like the problem is to do with the forwarding of RTP packets by Asterisk from A (Internet origin) to B (VPN). Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1)
Many thanks Tzafrir and Sergio, Now, I have another error when compiling zaptel : /lib/modules/2.6.8-2-686/build make -C /lib/modules/2.6.8-2-686/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernel-headers-2.6.8-2-686' CC [M] /usr/src/zaptel/zaptel.o In file included from include/asm/thread_info.h:16, from include/linux/thread_info.h:21, from include/linux/spinlock.h:12, from include/linux/capability.h:45, from include/linux/sched.h:7, from include/linux/module.h:10, from /usr/src/zaptel/zaptel.c:44: include/asm/processor.h:87: error: array type has incomplete element type /usr/src/zaptel/zaptel.c: In function '__zt_receive_chunk': /usr/src/zaptel/zaptel.c:6115: warning: pointer targets in assignment differ in signedness make[2]: *** [/usr/src/zaptel/zaptel.o] Erreur 1 make[1]: *** [_module_/usr/src/zaptel] Erreur 2 make[1]: Leaving directory `/usr/src/kernel-headers-2.6.8-2-686' make: *** [linux26] Erreur 2 sarge:/usr/src/zaptel# What to do more ? -Message d'origine- De : Sergio Serrano [mailto:[EMAIL PROTECTED] Envoyé : mardi 27 septembre 2005 09:36 À : [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : RE: [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1) You must install libncurses5-dev regards, Srsergio -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Tzafrir Cohen Envoyé : mardi 27 septembre 2005 09:33 À : asterisk-users@lists.digium.com Objet : Re: [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1) On Tue, Sep 27, 2005 at 09:20:13AM +0200, [EMAIL PROTECTED] wrote: Hello Gentlemen :-) I am a little disapointed by an error occured during an update from 1.0.7 to Head in a Debian testing distro. Start with defining a standard deb-src of Sarge (I think it is defined by default. Maybe remmed-out) and then run: apt-get install build-essential apt-get build-dep asterisk It should get you roughly the packages needed to build HEAD from source. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] blindxfer atxfer not working?
I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked Enter the number of packages, followed by the Pound key. I found on the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) that updates had been made to address this very problem, so I went to CVS HEAD and updated features.conf: [featuremap] blindxfer = *1; Blind transfer disconnect = *0 ; Disconnect atxfer = *2 ; Attended transfer Now, when a call comes in, I can press *1 and I hear Transfer, at which point I enter an extension and the call goes there. However if _I_ initiate the call, *1 does nothing - I cannot transfer the call. Same story for attended transfer (*2). It doesn't make any difference whether I place the call on a SIP or ZAP channel. Is this a bug? If not, what's the secret to transferring outgoing calls that I initiate? BTW, I have calltransfer=yes on ALL my ZAP channels and I'm using t in my Dial commands (I noticed that using T doesn't help – the called party can't transfer the call either). Thanks, Hugh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One-way audio with VPN
On Tue, September 27, 2005 20:22, Alex Lake said: I've got a one-way audio problem, but I've looked through a few documents on the subject and I'm not sure that it's the same issue. User A calls a local Asterisk user B via a public SIP gateway (voiptalk.org) using (sip:[EMAIL PROTECTED]) B is connected to the Asterisk server via VPN B is registered (and has successful bi-directional conversations with other users on the VPN) Asterisk correctly forwards the call via SIP and B's phone rings and is answered, but B can't hear A So there appears to be an audio-path blockage from A via Asterisk to B. Now if A leaves a voicemail message on the asterisk box, that's fine (the sound file contains a recording of A's voice!) Therefore, it looks like the problem is to do with the forwarding of RTP packets by Asterisk from A (Internet origin) to B (VPN). Any ideas? If you're not doing NAT on the SOURCE IP of the A before transferring across the VPN, it is very likely that B is replying DIRECTLY to A rather than through the VPN. This will cause B to answer with a different Source IP than A has initiated the call to, causing the packets to be dropped. You can easily check this by doing a packet trace on the LAN segment of B... Good luck! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 - problem dialing extra numbers
Change your dtmf setting. Covered lots of times before, or info on voip-info.com Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jachin Rupe Sent: Tuesday, 27 September 2005 1:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom IP 500 - problem dialing extra numbers hi there I'm setting up [EMAIL PROTECTED] and I'm using Polycom IP 500 phones. When I call a number that has a digital receptionist (i.e. dial 1 or such and such, dial 2 for this and that...) the Polycom doesn't seem to send the extra digits. When I try it with X-Lite things appear to work fine, so I think the problem is with the Polycom configuration. Here's some of my configuration files. If I didn't included an important one please let me know. - .cfg - ?xml version=1.0 standalone=yes? !-- Default Master SIP Configuration File-- !-- Edit and rename this file to Ethernet-address.cfg for each phone.-- !-- $Revision: 1.13 $ $Date: 2004/11/26 23:30:44 $ -- APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=phone1.cfg, sip.cfg MISC_FILES= LOG_FILE_DIRECTORY=/log/ / - sip.cfg - ?xml version=1.0 standalone=yes? !-- SIP Application Configuration File -- !-- $Revision: 1.63 $ $Date: 2004/11/08 18:52:16 $ -- sip voIpProt local voIpProt.local.port=/ server voIpProt.server.1.address=10.0.20.0 voIpProt.server.1.port=5060 voIpProt.server.1.transport=DNSnaptr voIpProt.server.1.expires=300 voIpProt.server.1.register=1 voIpProt.server.1.retryTimeOut=0 voIpProt.server.1.retryMaxCount=0 voIpProt.server.1.expires.lineSeize=30 / SIP voIpProt.SIP.useRFC2543hold=1 voIpProt.SIP.lcs=0 voIpProt.SIP.sendCompactHdrs=0 voIpProt.SIP.WM50=0 voIpProt.SIP.keepalive.sessionTimers=0 voIpProt.SIP.requestURI.E164.addGlobalPrefix= outboundProxy voIpProt.SIP.outboundProxy.address=10.0.20.0 voIpProt.SIP.outboundProxy.port=5060 / alertInfo voIpProt.SIP.alertInfo.1.value= voIpProt.SIP.alertInfo.1.class= / requestValidation voIpProt.SIP.requestValidation.1.request= voIpProt.SIP.requestValidation.1.method= voIpProt.SIP.requestValidation.1.request.1.event= digest voIpProt.SIP.requestValidation.digest.realm=PolycomSPIP / /requestValidation specialEvent voIpProt.SIP.specialEvent.lineSeize.nonStandard=1 voIpProt.SIP.specialEvent.checkSync.alwaysReboot=0/ conference voIpProt.SIP.conference.address= / /SIP /voIpProt dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap= dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address= dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan logging level change log.level.change.sip=4 log.level.change.sip.obs=5/ /level /logging /sip I just realized something... I don't have a phone1.cfg file, should I? I adopted this system in a partial working state from someone else and I'm still figuring out why things are the way they are. thanks -jachin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] analogue phone with asterisk
You need an Analog Terminal Adapter (ATA). Sipura makes some good ones. Check out http://www.voipsupply.com/product_info.php?cPath=96_118products_id=321 http://www.voipsupply.com/product_info.php?cPath=96_118products_id=713 That's what I use, and I love 'em. /edg --On Tuesday, September 27, 2005 12:10 PM -0500 Rajesh Bhairampally [EMAIL PROTECTED] wrote: I am a newbee to asterisk. I recently installed [EMAIL PROTECTED] Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a PSTN phone connection. I want to use my analogue phones as the end points for my asterisk box to make and receive calls. All i want is to use my analogue phones instead of soft phones. Can some one help me what hardware interface i need for that and how should i go about it? if there is any HOW-TO for that it will be of great help. thanks, rajesh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 500 - problem dialing extra numbers
Jachin Rupe wrote: hi there I'm setting up [EMAIL PROTECTED] and I'm using Polycom IP 500 phones. When I call a number that has a digital receptionist (i.e. dial 1 or such and such, dial 2 for this and that...) the Polycom doesn't seem to send the extra digits. When I try it with X-Lite things appear to work fine, so I think the problem is with the Polycom configuration. Sounds like it might be your DTMF handling. Try setting relaxdtmf=yes in your sip.conf, and if running cvs head, change dtmfmode=auto for sip peers if not using HEAD, then try dtmfmode=rfc2833 (g729 codec) or dtmfmode=inband (ulaw/alaw) do your debug logs show anything like double digits, or missing digits when you try calling IVR's? matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] analogue phone with asterisk
On Tue, September 27, 2005 19:10, Rajesh Bhairampally said: I am a newbee to asterisk. I recently installed [EMAIL PROTECTED] Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a PSTN phone connection. I want to use my analogue phones as the end points for my asterisk box to make and receive calls. All i want is to use my analogue phones instead of soft phones. Can some one help me what hardware interface i need for that and how should i go about it? if there is any HOW-TO for that it will be of great help. It depends on how many phones you have... You can start with a few IAXy's at the low end all the way to channel banks at the high end... The wiki is your friend: http://www.voip-info.org/tiki-index.php?page=Analog+Telephone+Adapters HTH -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Software only Asterisk PBX (commercial)
Don't you ever recommend Bicom as they take your money and will never deliver a product that works. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Morgan Gilroy Sent: Tuesday, September 27, 2005 1:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Software only Asterisk PBX (commercial) Also check out http://www.bicom.us pretty expensive but if that's your thing :) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ronald Hartmann Sent: 27 September 2005 16:47 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Software only Asterisk PBX (commercial) Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Have a look at the AMP project http://sourceforge.net/projects/amportal ~ron ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [MSG]TDM Error on ASUS Pundit-R
[EMAIL PROTECTED] wrote on 09/27/2005 01:18:35 PM: Hi I have looked around but I cant find an answer for this, I randomly get the error 'TDM PCI Master abort' and the system locks up. All I have found so far are a couple other posts on it but no solution. Running fedora core 3, asterisk stable, zaptel stable. I had this problem with a Rev F board. Upgrading to a newer board fixed it for me. I don't know if anyone has a more specific solution... Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [EMAIL PROTECTED] inbound call problem to SIP trunk. (voipfone UK)
Hi all, I have recently installed [EMAIL PROTECTED] and outbound calling is working great. However I am strugglings with inbound calls. I have setup a trunk for my provider, voipfone and in the inbound area on AMP I have the following :- user context name = 3011 context=from-pstn dtmfmode=rfc2283 fromdomain=voipfone.co.uk host=voipfone.co.uk insecure=very secret=XX type=user user=3011 username=3011 To be honest a lot of this is guesswork so could be wrong. I've tried a lot of others settings sut still get no inbound calls. I also went into inbound routing and created a default route with icoming calls sent to my extention. That is all I have done. If I call my PSTN number from the PSTN i get a log entry and it shows the calling PSTN number so It looks to me as though the trunk must be okay as the call is getting routed to my Asterisk, or am I mistaken with this? Does anyone know what Failed to authenticate user 0792124000 ;tag=as16492b07 means? Is it something to do with my inbound context? 07921 24000 is the PSTN number. here is the full log extract. p 27 14:30:53 DEBUG[2618] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 129: Match Found Sep 27 14:30:53 DEBUG[2618] chan_sip.c: Registration successful Sep 27 14:30:53 DEBUG[2618] chan_sip.c: Cancelling timeout 14095 Sep 27 14:31:03 DEBUG[2618] chan_iax2.c: Peer lastms 33, historicms 33, maxms 2000 Sep 27 14:31:09 DEBUG[2618] manager.c: Manager received command 'Command' Sep 27 14:31:09 DEBUG[2618] manager.c: Manager received command 'Command' Sep 27 14:31:09 DEBUG[2618] manager.c: Manager received command 'MailboxStatus' Sep 27 14:31:25 DEBUG[2618] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Sep 27 14:31:27 DEBUG[2618] chan_sip.c: Setting NAT on RTP to 0 Sep 27 14:31:27 DEBUG[2618] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 102: Match Found Sep 27 14:31:27 DEBUG[2618] chan_sip.c: Setting NAT on RTP to 0 Sep 27 14:31:27 NOTICE[2618] chan_sip.c: Failed to authenticate user 07921249135 ;tag=as16492b07 Sep 27 14:31:27 DEBUG[2618] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 103: Match Found Sep 27 14:31:38 DEBUG[2618] chan_sip.c: Scheduled a registration timeout for voipfone.co.uk id #14103 Sep 27 14:31:38 DEBUG[2618] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 130: Found Sep 27 14:31:38 DEBUG[2618] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 130: Match Found Sep 27 14:31:38 DEBUG[2618] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 131: Found Sep 27 14:31:38 DEBUG[2618] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 131: Match Found Sep 27 14:31:38 DEBUG[2618] chan_sip.c: Registration successful Sep 27 14:31:38 DEBUG[2618] chan_sip.c: Cancelling timeout 14103 Sep 27 14:31:38 DEBUG[2618] chan_sip.c: Cancelling timeout 14103 I've tried for a week now and could really use some help! Thanks Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FSX/UK analogue Phone rings all the time
Best thing is to get a 'Master' or PBX Master socket, cut one end off an RJ11-RJ11 lead, and connect the red/green pair (centre two pins or the RJ11) to pins 2 and 5 of a master socket. Rgds Tim John Crowhurst wrote: On Mon, September 26, 2005 20:35, Asterisk said: hi Asterisk users, I am in the UK and trying to get an asterisk system running. I have the SIP side of things running or limping along to the best of my newbie ability. I have a problem with a FXS card. Connecting a standard (Working) UK phone makes the phone ring all the time while on hook. Sounds like the A/B is being coupled onto the ring wire. I've heard somewhere that you need to connect the phone through a master socket. I think its something to do with the ringing signal from the FXS card. -- John ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Tandem Inbound only.
Hello, I have a carrier that is supplying me with DID inbound over SIP to my asterisk server. Because the CID is different with every call that is coming in the only way I have to authenticate this carrier is IP based. In my sip.conf I want to define this user as type=user, however this can't work because Asterisk only authenticates users by username, not IP. I can take calls in if I set type=friend or type=peer which will allow authentication by IP. The problem with this is that asterisk sends sip OPTIONS messages to the carrier, because asterisk thinks that the carrier will be receiving calls as well as sending calls. The options messages make the carrier very unhappy, and just throw errors on their end. I believe that if I were to put Ser in front of asterisk it would resolve this issue, but that seems a bit drastic, and it is not justified I think. Does anyone have an idea of how to stop asterisk from sending options messages to peers or friends, or how to authenticate based on IP address for a user? Thanks, - Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What ISDN hardware would you recommend?
Quad or octo-bri from www.junghanns.net We use a few of these and they are not cheap but they work without any hassle. Rgds Tim Robinson Francesco Peeters wrote: The machines themselves will not pose much of a problem, but what ISDN hardware would you recommend for this? (1 site with 1 TE and 1 NT mode port, 2 sites with 2 TE and 2 NT mode ports) TIA! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Tandem Inbound only.
On 27/09/05, Scott Eisert [EMAIL PROTECTED] wrote: Hello, I have a carrier that is supplying me with DID inbound over SIP to my asterisk server. Because the CID is different with every call that is coming in the only way I have to authenticate this carrier is IP based. In my sip.conf I want to define this user as type=user, however this can't work because Asterisk only authenticates users by username, not IP. Check out 'insecure=very' for sip.conf. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Best drivers for HFC-S ISDN cards
Hi As requested here are my configs. I have 3 zaphfc cards - 2 in NT mode and 1 in TE mode connected to the BT network. I have a variety of phones - a Cisco 7940, a Snom 190, a Grandsteam Budgie plus 2 cordless ISDN phones on one of the NT ports, and a Network Alchemy Cybergear Gold on the other connected to some analogue phones. no echo. nil. niets! Please post a diagram of your system config and we can take a look at it if you need some help. Rgds Tim Here is zaptel.conf loadzone=uk defaultzone=uk span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,0,3,ccs,ami bchan=4-5 dchan=6 span=3,0,3,ccs,ami bchan=7-8 dchan=9 Here is my zapata.conf file ; ; Zapata telephony interface ; ; Configuration file ;NT mode - extension card [channels] nocid=Unavailable withheldcid=Withheld language=en usecallerid=yes callwaiting=yes nationalprefix=0 internationalprefix=00 switchtype = euroisdn signalling = bri_net_ptmp pridialplan=local prilocaldialplan=local echocancel=yes echocancelwhenbridged=no immediate=no overlapdial=yes group = 1 context=cybergear-in channel = 1-2 ;NT mode - extension card nocid=Unavailable withheldcid=Withheld language=en usecallerid=yes callwaiting=yes nationalprefix=0 internationalprefix=00 switchtype = euroisdn signalling = bri_net_ptmp pridialplan=local prilocaldialplan=local echocancel=yes echocancelwhenbridged=no immediate=no overlapdial=yes group = 2 context=isdn-phones-in channel = 4-5 ;TE mode - for ISDN line nocid=Unavailable withheldcid=Withheld Language=en usecallerid=yes pridialplan=unknown nationalprefix=0 internationalprefix=00 switchtype = euroisdn signalling = bri_cpe_ptmp echocancel=yes echocancelwhenbridged=no immediate=no overlapdial=yes group = 3 context=isdn-in channel = 7-8 Giordano Grandis wrote: Well done Tim...could u post here your Zapata.conf ? :) I'm in Italy and have some issues with echo Thanks Giordano Grandis [EMAIL PROTECTED] -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Tim Robinson Inviato: lunedì 26 settembre 2005 22.30 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Best drivers for HFC-S ISDN cards Chris I have only ever used zaphfc drivers and for me they are perfect. Echo has never been a problem. It would be helpful if you were to provide a bit more information to the group about your configuration so we can try and help you work out the cause. Switching to capi or mISDN is unlikely to help and will almost certainly be a retrograde step as far as I hear from these forums. Best regards Tim Robinson Basingstoke, UK Chris Bagnall wrote: It seems that HFC-S cards can be connected with asterisk in a few different ways - isdn4linux, mISDN, chan_capi or zaphfc. I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). What's the recommended way to hook up these ISDN cards? Is switching to capi or mISDN likely to remove the echo problem completely, or is this one of those things one has to accept? Thanks in advance. Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] blindxfer atxfer not working?
double-check your usage of the t and T parameters to the Dial command, detailed here: http://www.voip-info.org/wiki-Asterisk+cmd+Dial Mojo hugolivude wrote: I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked Enter the number of packages, followed by the Pound key. I found on the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) that updates had been made to address this very problem, so I went to CVS HEAD and updated features.conf: [featuremap] blindxfer = *1; Blind transfer disconnect = *0 ; Disconnect atxfer = *2 ; Attended transfer Now, when a call comes in, I can press *1 and I hear Transfer, at which point I enter an extension and the call goes there. However if _I_ initiate the call, *1 does nothing - I cannot transfer the call. Same story for attended transfer (*2). It doesn't make any difference whether I place the call on a SIP or ZAP channel. Is this a bug? If not, what's the secret to transferring outgoing calls that I initiate? BTW, I have calltransfer=yes on ALL my ZAP channels and I'm using t in my Dial commands (I noticed that using T doesn't help – the called party can't transfer the call either). Thanks, Hugh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best drivers for HFC-S ISDN cards
Chris Looking at this file below you need to move the stuff below the channel = 4-5 line to each span definition. Anything below the channel line gets completely ignored! I stand to be corrected but I think your current config will not have any echo cancellation at all. I have just posted my own zapata.conf files to this list. Take a look. Rgds Tim Robinson Basingstoke, UK Chris Bagnall wrote: I said: I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). Many thanks to those who replied. General consensus seems to be switching to mISDN or CAPI won't solve the intermittent echo problem. A follow-up with some more config information: Zaptel.conf span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,3,ccs,ami bchan=4-5 dchan=6 loadzone=uk defaultzone=uk Zapata.conf: [channels] language = en pridialplan = dynamic nationalprefix = 0 internationalprefix = 00 switchtype = euroisdn signalling=bri_cpe_ptmp group=1 context=isdn channel = 1-2 switchtype = euroisdn signalling=bri_cpe_ptmp group=2 context=isdn channel = 4-5 echocancel=yes echocancelwhenbridged=yes echotraining=no ;echotraining=800 rxgain=0.0 txgain=0.0 immediate=yes I've tried echotraining off, on, 100, 400 and 800, none of which seem to help matters very much. Any suggestions for getting rid of echo on these lines would be gratefully appreciated. Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Setup Questions
So while I'm waiting to see if anyone can help with those questions, I thought I would ask one more :-) All of the sudden 3 of my Polycom501 handsets started having a 1 way audio problem. My setup: 30 Polycom501 handsets all connected to Asterisk (CVS HEAD from a week or two ago) over a 100Mb LAN. The * server isn't using a firewall. What Happened: The 3 phones in question were working morning yesterday, then for no apparent reason, the user could no longer talk. The polycom user could hear the person at the other end, but could not talk to them. Nothing has changed as far as I can tell, and I have no idea even where to start looking. Also I did the on phone Diagnositics and the handset is working according to that. Any help would be greatly appreciated. Thanks, Matthew O'Connor Matthew T. O'Connor wrote: OK I have just gone live with asterisk in a new office with approx 40 Polycom 501 handsets. I have a few questions: 1) Call Parking: I am able to park calls using the standard Asterisk call parking system (transfer to ext *70 etc...) I would like to make this a little easier for my users The Polycom 501s w/ SIP 1.5.2 seem to support some type of standard call parking, however I don't think it works with Asterisk. Is this true? Is there a way to integrate the to call parking system etc? 1a) If I can't use the Polycom built-in call park feature, is there a way to remap one of the buttons on the left (the services button for example) to dial *70 for my users? 2) Transferring Calls: They way our office operates, I would prefer the default transfer method to be a blind transfer. Is there a way to reprogram the Polycoms to default to blind transfers? There are more questions but that is all for now :-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Review: Digium TE405P v2
Hello, We have finished our tests of the new Digium firmware on the quad T1 cards(TE405P/TE410P). Overall it is a big improvement over the version 1 firmware. Here's the review: http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html MATT--- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Tandem Inbound only.
On Tuesday 27 September 2005 3:12 pm, Peter Bowyer wrote: On 27/09/05, Scott Eisert [EMAIL PROTECTED] wrote: Hello, I have a carrier that is supplying me with DID inbound over SIP to my asterisk server. Because the CID is different with every call that is coming in the only way I have to authenticate this carrier is IP based. In my sip.conf I want to define this user as type=user, however this can't work because Asterisk only authenticates users by username, not IP. Check out 'insecure=very' for sip.conf. Peter It doesn't look like insecure can solve my problem. If I have type=user, I send back a 404 regardless of the insecure setting. If I have type=peer or type=friend I can receive calls but asterisk sends out Options messages regardless of the insecure setting (yes or very). Any other suggestions? - Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cgi-bin/vmail.cgi - - Invalid Context
Greetings: I have been playing around with vmail.cgi and am able to log into and listen to my message with no problem. I added the correct context to vmail.cgi so I don't have to enter the mailbox + context. However, when I try and delete a message or move to a different mailbox I get the following: Code: Software error: Invalid ContextBR Running * v1.0.8 Any help would be appreciated ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Tandem Inbound only.
Hi Scott, To do what you want to do you do indeed need to use a peer entry, with the IP address where INVITEs will come from specified as the host, and insecure=very. Your OPTIONS though is being caused by qualify being turned on somewhere. Joshua Colp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Eisert Sent: Tuesday, September 27, 2005 4:48 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Tandem Inbound only. On Tuesday 27 September 2005 3:12 pm, Peter Bowyer wrote: On 27/09/05, Scott Eisert [EMAIL PROTECTED] wrote: Hello, I have a carrier that is supplying me with DID inbound over SIP to my asterisk server. Because the CID is different with every call that is coming in the only way I have to authenticate this carrier is IP based. In my sip.conf I want to define this user as type=user, however this can't work because Asterisk only authenticates users by username, not IP. Check out 'insecure=very' for sip.conf. Peter It doesn't look like insecure can solve my problem. If I have type=user, I send back a 404 regardless of the insecure setting. If I have type=peer or type=friend I can receive calls but asterisk sends out Options messages regardless of the insecure setting (yes or very). Any other suggestions? - Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Setup Questions
Matthew T. O'Connor wrote: The 3 phones in question were working morning yesterday, then for no apparent reason, the user could no longer talk. The polycom user could hear the person at the other end, but could not talk to them. Nothing has changed as far as I can tell, and I have no idea even where to start looking. Could be one of three things: 1 - Codec Problems 2 - NAT (but you mention no firewall and same LAN, so not this one) 3 - Polycom's known problems I'm willing to bet 3. Try rebooting the phone and seeing if it works for you then. The wiki has documentation on this if you're not sure how to reboot them. Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Setup Questions
At 14:40 9/27/2005, Matthew T. O'Connor, wrote: So while I'm waiting to see if anyone can help with those questions, I thought I would ask one more :-) All of the sudden 3 of my Polycom501 handsets started having a 1 way audio problem. Did you make certain canreinvite equals no? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Software only Asterisk PBX (commercial)
I would take a look at Signate, too. Tom On Sep 27, 2005, at 11:12 AM, Matthew Crocker wrote: Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AstriCon 2005 - Now With Free Beer!
AstriCon Update: Only Two Weeks To Go! October 12 - 14, 2005 Anaheim, CA AstriCon 2005 starts two weeks from today. We now have a complete roster of speakers covering Asterisk from soho to carrier. We've added the Code Zone, a working lab with a full compliment of VoIP and TDM equipment. We also have over 20 confirmed exhibitors and more are joining the event each day. AstriCon 2005 is shaping up to be a three day Asterisk extravaganza. AstriCon 2005 Highlights: - Keynotes from Asterisk Leaders: * Mark Spencer: Asterisk 1.2 And Beyond * Carrier Grade, Fault-Tolerant Asterisk * Asterisk VoIP Emergency Call Handling - Asterisk 1.2: Enhancements, Features Changes - Free copy of Asterisk: The Future of Telephony from O'Reilly * By Jim VanMeggalen, Leif Madsen, Jared Smith * Free to the first 500 tutorial attendees - 3 SIG Tracks: * Enterprise Asterisk * Call Center Operators * ITSPs Carriers - The Asterisk Expo: 20+ Asterisk Related Vendors - The Open Source Showcase: Asterisk-related open source projects * AstLinux * AsterNIC * Asterisk on WRT54G * astGUIclient/VICIDIAL * Zap Radio - 3 Tutorial Tracks: * Beginners * Intermediate Advanced * Developer - The Code Zone: a lab stocked with hardware and Red Bull * Come in and code! * Meet the gurus of Asterisk. * Test out solutions * Show off your code - Huge Party: The Golden Asterisk Pub * J.T. Schmidt's Brewery * Free Beer (Thanks Digium!) Register today: http://www.astricon.net/2005/ Don't forget that space at AstriCon 2005 is limited. Please register as soon as possible to insure admission to your preferred tutorial and conference tracks. Hotel space is also limited. Reserve your room today: http://www.astricon.net/2005/hotel.shtml Please contact us if you have any questions: [EMAIL PROTECTED] or by phone at +1 816 256 8916. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] wait before accepting the call
Why don't you write a couple of lines AGI scripts that will call asterisk command WAIT(5) Thankx -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 27 Sep 2005 13:42:31 +0200 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] wait before accepting the call hello! i'm looking for a way to prolonge a pstn-call for 5 seconds before it enters the extensions.conf. this is for testing purposes, all numbers of a ddi should be received by asterisk before the call is walking through the extensions. how can i achive this? i've not seen a feature like this for zapata or zaptel, does anyone have an idea how this could be done? thx christian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Subject: [Asterisk-Users] Vonage-type service
On Monday 26 September 2005 10:41, Federico Alves wrote: We don't sell the system. We provide a full independent system for customers including co-location, for a setup fee and 1/2 cent per call, regardless of length. We also provide US termination via our own DS3 for 1.3 cents a minute, and it does support T.38 faxing. I believe this belongs on [EMAIL PROTECTED] -- José Pablo Ezequiel Fernández ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Setup Questions
I had a loose headset cable doing that one day -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: Tuesday, September 27, 2005 3:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom Setup Questions Matthew T. O'Connor wrote: The 3 phones in question were working morning yesterday, then for no apparent reason, the user could no longer talk. The polycom user could hear the person at the other end, but could not talk to them. Nothing has changed as far as I can tell, and I have no idea even where to start looking. Could be one of three things: 1 - Codec Problems 2 - NAT (but you mention no firewall and same LAN, so not this one) 3 - Polycom's known problems I'm willing to bet 3. Try rebooting the phone and seeing if it works for you then. The wiki has documentation on this if you're not sure how to reboot them. Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
