[Asterisk-Users] doubling digits
I am not sure if anyone is having this same problem, but, with * as the IVR on 2 long distance T1's that we have, serving some 16000 customers, as they enter their phone numbers or any other group of digits, some number get doubled.. example: 773-259-2019 might be picked up as 773-255-9201... I heard many complaints. We have relaxdtmf = yes. We don't have PRI. I'm not sure if that will solve it. Most of the input is being picked up by AGI (in perl). Also note, within 3 months the 2 T1's are going to be replace by a DS3.. so, I really need some help on this one before we go with more lines.. thanks Omar Abhari PlatinumTel Communications ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question on soft phones with SIP
Nice work Andy. Filled in some issues I didn't quite understand. Looking forward to more. Chris Tielman, You can take a look at the quick and dirty guide I'm slowly putting together if you like... http://www.automated.it/guidetoasterisk.htm I'd appreciate any feedback you have on it.. and if it helped Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue App Patch
Hi there, I am not a true C programmer. PERL is my game, just hacked around at C for a long time. Anyways, I digress... Here is a patch, can you tell me: 1) Can you see any blaring security holes I may have created 2) Does it make sense. It is for the queue app. It will allow you to do a few things (currently not finished :) 1) Set an announcetimeout in queues.conf. This variable seemed unused in app_queue.c This is the time in seconds in which I will stop MOH, and play something like You are currently caller # XXX 2) I also have planned for small advertisements to cut into the MOH, I am thinking of a separate directory, and playing a random advertisement every so often, hence the advert_timeout and advert_dir in queues.conf. Please give me feedback, this is my first release of a patch to ANY community, I want to know how badly I suck :). Everyone that I have talked to in the asterisk community has been GREAT! Thank you, John Congdon congdonj (IRC) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue App Patch, addendum
Another thing I am working on is to do the timeout people have been asking about. If they have been on hold for (X minutes) dump out of the Queue App, so that the next priority can be a common mailbox. And I forgot to add the patch to my last email!!! argh http://pbx.usedontmiss.com/queue_patch John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VON in London..
200 bucks On Saturday 07 June 2003 1:46 pm, WipeOut . wrote: Hey, Anyone have any idea what the cost to visit the VON exhibition in London next week? Not the Conference thats just way to far out of my budget.. Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New cdr_mysql.c
Hi.
here is a brand new cdr_mysql .
Is based on the previous one, but with lots of changes, so I include
here the whole file, not a patch.
What I've changed:
added new fields, in order to reflect cdr csv :
* call start time
* call answer time
* call end time
* call unique id
Changed the table structure to reflect cdr.h lengths.
Added some (a lot?) sanity checks, to be sure to insert
records that fits into mysql, and not to exit with Failed insert
errors. Also now on load/reload cdr mysql checks the table structure
to adjust the fields length values (and issue warnings if some
fields are too small).
Feel free to blame me (I'm not a C guru). I can fix any error, of
course. Mark feel free to add it to asterisk, if u think that's
a good work. I'll disclaim, if needed.
Positive and/or negative comments are welcome.
Matteo.
--
Matteo Brancaleoni
Powered by RedHat Linux 8.0
Linux User #153521
-BEGIN GEEK CODE BLOCK-
Version: 3.12
GS d? s:- a- C+++ UL P+ L+++ E- W+++ N++ o K- w--
O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+
G e h! r++ y
--END GEEK CODE BLOCK--
# phpMyAdmin MySQL-Dump
# version 2.5.0
# http://www.phpmyadmin.net/ (download page)
#
# Host: localhost
# Generato il: 07 Giu, 2003 at 03:23
# Versione MySQL: 3.23.56
# Versione PHP: 4.2.2
# Database : `asterisk`
#
#
# Struttura della tabella `cdr`
#
# Creazione: 07 Giu, 2003 at 02:57
# Ultimo cambiamento: 07 Giu, 2003 at 03:07
#
CREATE TABLE `cdr` (
`ID` int(10) unsigned NOT NULL auto_increment,
`calldate` datetime NOT NULL default '-00-00 00:00:00',
`clid` varchar(80) NOT NULL default '',
`src` varchar(80) NOT NULL default '',
`dst` varchar(80) NOT NULL default '',
`dcontext` varchar(80) NOT NULL default '',
`channel` varchar(80) NOT NULL default '',
`dstchannel` varchar(80) NOT NULL default '',
`lastapp` varchar(80) NOT NULL default '',
`lastdata` varchar(80) NOT NULL default '',
`start` datetime NOT NULL default '-00-00 00:00:00',
`answer` datetime NOT NULL default '-00-00 00:00:00',
`end` datetime NOT NULL default '-00-00 00:00:00',
`duration` int(11) NOT NULL default '0',
`billsec` int(11) NOT NULL default '0',
`disposition` int(11) NOT NULL default '0',
`amaflags` int(11) NOT NULL default '0',
`accountcode` varchar(20) NOT NULL default '',
`uniqueid` varchar(32) NOT NULL default '',
PRIMARY KEY (`ID`)
) TYPE=MyISAM AUTO_INCREMENT=2 ;
/*
* Asterisk -- A telephony toolkit for Linux.
*
* MySQL CDR logger
*
* Matteo Brancaleoni [EMAIL PROTECTED]
* based on the first version by
* James Sharp [EMAIL PROTECTED]
*
* This program is free software, distributed under the terms of
* the GNU General Public License.
*
*
* Table Structure for `cdr`
*
* Created on: 07 Giu, 2003 at 02:57
* Last changed on: 07 Giu, 2003 at 03:07
CREATE TABLE `cdr` (
`ID` int(10) unsigned NOT NULL auto_increment,
`calldate` datetime NOT NULL default '-00-00 00:00:00',
`clid` varchar(80) NOT NULL default '',
`src` varchar(80) NOT NULL default '',
`dst` varchar(80) NOT NULL default '',
`dcontext` varchar(80) NOT NULL default '',
`channel` varchar(80) NOT NULL default '',
`dstchannel` varchar(80) NOT NULL default '',
`lastapp` varchar(80) NOT NULL default '',
`lastdata` varchar(80) NOT NULL default '',
`start` datetime NOT NULL default '-00-00 00:00:00',
`answer` datetime NOT NULL default '-00-00 00:00:00',
`end` datetime NOT NULL default '-00-00 00:00:00',
`duration` int(11) NOT NULL default '0',
`billsec` int(11) NOT NULL default '0',
`disposition` int(11) NOT NULL default '0',
`amaflags` int(11) NOT NULL default '0',
`accountcode` varchar(20) NOT NULL default '',
`uniqueid` varchar(32) NOT NULL default '',
PRIMARY KEY (`ID`)
) TYPE=MyISAM AUTO_INCREMENT=2 ;
*/
#include sys/types.h
#include asterisk/config.h
#include asterisk/options.h
#include asterisk/channel.h
#include asterisk/cdr.h
#include asterisk/module.h
#include asterisk/logger.h
#include ../asterisk.h
#include stdio.h
#include string.h
#include stdlib.h
#include unistd.h
#include time.h
#include math.h
#include mysql.h
#define DATE_FORMAT %Y-%m-%d %T
static char *desc = MySQL CDR Backend;
static char *name = mysql;
static char *config = cdr_mysql.conf;
static MYSQL *mysql;
static MYSQL_RES *mysql2;
/* some defaults, based on cdr.h */
static int calldate=19;
static int start=19;
static int end=19;
static int answer=19;
static int clid=80;
static int src=80;
static int dst=80;
static int dcontext=80;
static int channel=80;
static int dstchannel=80;
static int lastapp=80;
static int lastdata=80;
static int accountcode=20;
static int uniqueid=32;
static int duration=11;
static int billsec=11;
static int disposition=11;
static int amaflags=11;
static int mysql_log(struct ast_cdr *cdr)
{
struct tm tm;
struct timeval tv;
struct timezone tz;
char *sqlcmd, timestr[128],
[Asterisk-Users] SIP, NAT Asterisk
Hi all, beacause I am a newbie in the asterisk ralm and the existing documentation could not satisfy I'd like to ask you some Questions: 1. Does somewhere in the Internet exist additional documentations for asterisk configuration ? 2. Does Asterisk work as a standard SIP Proxy ? 3. I am just installing a Asterisk PBX in our institute and additionally I purchased some ot the Snom 100 SIP devices. For SIP account I have registered with iptel.org and fwd.pulver.com. Both work fine with the Snom 100 device. I tried the Snom 100 from home via a Linksys DSL router wich supports UPnp and port range forwarding to the private network. I faced some troubles with registering the Snom with IPTEL.ORG from my NAT'ed network. Because I'd like to setup a Asterisk PBX to my private Network as well my approach is to connect the SNOM to the asterisk at home network and establish a server-server connection via IAX to the institutes Asterisk server. The institute's Asterisk can login at iptel.org and fwd.pulver.com. If soembody wants to call me he should first try to reach me at my Snom phone at institutes premisses and after a defined time this call should be forwarded to my home asterisk server to reach the SNom 100 at home. BTW - for calling out I want to use the AVM Fritz ISDN card with ISDN4LINUX. I could not find any doc for Asterisk- ISDN4LINUX configuration. Later I'd like to upgrade to a Digium device to support conferencing. Wich of the Digium's devices you propose for a normal EuroISDN line (2 B-channels) ? I have attached a small figure for my SIP configuration aaproach. Thank you for your help regards Olaf -- Dipl. Ing. Olaf Menzel - System Engineer FOKUS - Fraunhofer Institute for Open Communication Systems: - Competence Center for Advanced Satellite Communication Schloss Birlinghoven, 53754 Sankt Augustin, Germany Phone: +49-2241-14-3494 Mobile: +49-175-2616161Fax: +49-2241-14-43494 email: [EMAIL PROTECTED] Internet: http://www.fokus.fhg.de/satcomattachment: asterisk-conf.gif
Re: [Asterisk-Users] doubling digits
I think that most of the companies that have IVR calling cards applications experience similar problem. It's because of the cheap phones some customers use or because of the noise on the line. So when they press a DTMF digit the generated tone is interrupted by some short noise on the line and the detector sees another digit (especially on compressed circuits) With that said you propably need a live sample that caused the problem. That could be done using the Monitor application and then when you hear about the complaint you can manage to find the right wave file and see exactly what happened. regards Martin On Sat, 7 Jun 2003, Omar Abhari wrote: I am not sure if anyone is having this same problem, but, with * as the IVR on 2 long distance T1's that we have, serving some 16000 customers, as they enter their phone numbers or any other group of digits, some number get doubled.. example: 773-259-2019 might be picked up as 773-255-9201... I heard many complaints. We have relaxdtmf = yes. We don't have PRI. I'm not sure if that will solve it. Most of the input is being picked up by AGI (in perl). Also note, within 3 months the 2 T1's are going to be replace by a DS3.. so, I really need some help on this one before we go with more lines.. thanks Omar Abhari PlatinumTel Communications ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] install asterisk without FXO PCI or modem? Isit possible! TXT FILE NOW!
Or compile for PROC=i586 in asterisk/Makefile Martin On Sat, 7 Jun 2003, Gary wrote: try putting in modules.con noload = ?? On Sat, 07 Jun 2003 00:04:56 -0400, hallian hallian wrote: Hello all - This is my situation! I have a PC with no PCI slot and no modem! But I would like to install asterisk on it. I want to use SIP based software ONLY with the asterisk PBX system. I already have the asterisk running on a regular PC. BUt when I run assterisk it fails at this point when it parse modules.conf file. [Wait] == Registered application 'Wait' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so]Illegal instruction I have played around with the file and have realzied if it runs successfully the chan_modem.so, then all other l;oadable modules are loaded. I even went to the extent of saying load module=no and copied/paste all the modules on the file! You know when are testing just maybe it might work! :-) But how can i load all my loadadble modules and skip this chan_modem.so option from the modules.conf file? Is it even possible? Thanks hallian _ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP, NAT Asterisk
Hi all, beacause I am a newbie in the asterisk ralm and the existing documentation could not satisfy I'd like to ask you some Questions: 1. Does somewhere in the Internet exist additional documentations for asterisk configuration ? http://www.digium.com/handbook-draft.pdf In addition, there are a variety of home-built pages. http://www.automated.it/guidetoasterisk.htm http://asterisk.gnuinter.net/ 2. Does Asterisk work as a standard SIP Proxy ? No. Asterisk can perform basic SIP redirection, but it is not a standard SIP proxy in many of the ways that you might expect. It can do _some_ of the features of a SIP proxy, but to call it a SIP proxy would be an overstatement. 3. I am just installing a Asterisk PBX in our institute and additionally I purchased some ot the Snom 100 SIP devices. For SIP account I have registered with iptel.org and fwd.pulver.com. Both work fine with the Snom 100 device. I tried the Snom 100 from home via a Linksys DSL router wich supports UPnp and port range forwarding to the private network. I faced some troubles with registering the Snom with IPTEL.ORG from my NAT'ed network. Because I'd like to setup a Asterisk PBX to my private Network as well my approach is to connect the SNOM to the asterisk at home network and establish a server-server connection via IAX to the institutes Asterisk server. The institute's Asterisk can login at iptel.org and fwd.pulver.com. If soembody wants to call me he should first try to reach me at my Snom phone at institutes premisses and after a defined time this call should be forwarded to my home asterisk server to reach the SNom 100 at home. BTW - for calling out I want to use the AVM Fritz ISDN card with ISDN4LINUX. I could not find any doc for Asterisk- ISDN4LINUX configuration. Later I'd like to upgrade to a Digium device to support conferencing. Wich of the Digium's devices you propose for a normal EuroISDN line (2 B-channels) ? I have attached a small figure for my SIP configuration aaproach. My suggestion is that if your Asterisk server is behind a NAT, the only hosts it should talk with are devices that are also behind the NAT, or other Asterisk servers via IAX or IAX2 which may be on the outside of the NAT. Anything else causes more headaches than it's worth. I'm afraid I have no experience with the EuroISDN equipment, so I am not qualified to answer that part of your questions. JT Thank you for your help regards Olaf -- Dipl. Ing. Olaf Menzel - System Engineer FOKUS - Fraunhofer Institute for Open Communication Systems: - Competence Center for Advanced Satellite Communication Schloss Birlinghoven, 53754 Sankt Augustin, Germany Phone: +49-2241-14-3494 Mobile: +49-175-2616161Fax: +49-2241-14-43494 email: [EMAIL PROTECTED] Internet: http://www.fokus.fhg.de/satcom Attachment converted: PrivateSpace:asterisk-conf.gif (GIFf/prvw) (000324C8) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] doubling digits
But see, we are not an IVR calling card company. It's a prepaid wireless gig. There is no transferring or what not. It's all simple DB transactions. All the cellphones that customers call from are digital TDMA phones, very good quality, mostly new. Land line, or cell phone, the problem still persists. Martin Pycko wrote: I think that most of the companies that have IVR calling cards applications experience similar problem. It's because of the cheap phones some customers use or because of the noise on the line. So when they press a DTMF digit the generated tone is interrupted by some short noise on the line and the detector sees another digit (especially on compressed circuits) With that said you propably need a live sample that caused the problem. That could be done using the Monitor application and then when you hear about the complaint you can manage to find the right wave file and see exactly what happened. regards Martin On Sat, 7 Jun 2003, Omar Abhari wrote: I am not sure if anyone is having this same problem, but, with * as the IVR on 2 long distance T1's that we have, serving some 16000 customers, as they enter their phone numbers or any other group of digits, some number get doubled.. example: 773-259-2019 might be picked up as 773-255-9201... I heard many complaints. We have relaxdtmf = yes. We don't have PRI. I'm not sure if that will solve it. Most of the input is being picked up by AGI (in perl). Also note, within 3 months the 2 T1's are going to be replace by a DS3.. so, I really need some help on this one before we go with more lines.. thanks Omar Abhari PlatinumTel Communications ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bandwidth measurement tool: bmtools
This is not specifically on-topic for Asterisk, but I have found on many occasions while working with Asterisk that it would have been very handy to be able to measure, with some precision, the bandwidth being used by a particular host, port, or combination of the two. So, I went searching for various tools, none of which were what I wanted. They either were too clever, or too limited in their abilities. However, someone forwarded the link to this tool to me about an hour ago, and I've been thrilled that it does _exactly_ what I want. I can use a BPF-style filter to monitor exactly what I'd like to watch, and it hands back results to me in real time down to a one-second interval. Sometimes, a small program can make me very happy, and I suppose after a morning full of various system problems I'm overly happy have something that works and does just what I want it to. This is useful for checking to see how much bandwidth a codec _really_ uses, or seeing what your total usage is between two IAX hosts, or pretty much anything that requires live examination of ethernet segment traffic. http://s-tech.linux-pl.com/bmtools/ [EMAIL PROTECTED] bmtools-0.71]# ./rate -r 1 -f 'host 10.0.1.3 and not port ssh' - Currently 263.05 Bps/3.01 pps, Average: 263.05 Bps/3.01 pps - Currently 2706.00 Bps/17.00 pps, Average: 1486.97 Bps/10.02 pps - Currently 588.00 Bps/6.00 pps, Average: 1186.92 Bps/8.68 pps - Currently 440.00 Bps/4.00 pps, Average: 1000.00 Bps/7.51 pps - Currently 440.00 Bps/4.00 pps, Average: 887.91 Bps/6.81 pps - Currently 2080.00 Bps/16.00 pps, Average: 1086.72 Bps/8.34 pps - Currently 1282.00 Bps/9.00 pps, Average: 1114.64 Bps/8.43 pps - Currently 10385.00 Bps/20.00 pps, Average: 2274.01 Bps/9.88 pps ^C JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip channel driver causes asterisk to crashwhen talking to quintum A800
On Sat, 7 Jun 2003, shido wrote: This is the sip debug when the call went through Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Content-Length: 157 Content-Type: application/sdp CSeq: 1 INVITE From: sip:[EMAIL PROTECTED];tag=402ada92-5 To: sip:[EMAIL PROTECTED] User-Agent: Quintum/1.0.0 Via: SIP/2.0/UDP 64.42.218.146;branch=z9hG4bK-tenor-64.42.218.146-5 Quintum: 0c01030b0239380501 v=0 o=Quintum 4 4 IN IP4 64.42.218.146 s=VoipCall c=IN IP4 64.42.218.146 t=0 0 m=audio 10240 RTP/AVP 0 c=IN IP4 64.42.218.146 a=rtpmap:0 pcmu/8000/1 11 headers, 8 lines Using latest request as basis request Sending to 64.42.218.146 : 5060 (non-NAT) Capabilities: us - 4, them - 4, combined - 4 Non-codec capabilities: us - 1, them - 0, combined - 0 Funnily enough I've been looking at the same problem. Will get a chance to look a bit more tomorrow. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wish-to-have in Asterisk
Dear Pals I think all of us in this business (or most) looking at the Corn Flakes Back Side , Wish to find a NAT Friendly VOIP solution. Thats ridiculous I know, but what about this wish list (Personal Point of view) Upnp support for Asterisk STUN Support for Asterisk Small Embedded Asterisk (just to redirect IAX) SIP-IAX , h.323-IAX or something like that , on the End Point. As a Nat friendly end point the most diffuctult Integrate IAX to RFCXX SIP , or H.323 to be recognized as Standard. Go to SIP Interop forums and look at the possibility to integrate IAX to SIP( or something like that) Manufacturers include IAX as Sip, MGCP, H.323 Mark and all the others, May be a good idea to start a WISH LIST for asterisk? Not for the experts in development, Just let comon Asterisk Users (Like Me) to put my bit to this interesting proyect. So if there`s a specialized developer get his Wish-List Point and start over and move forward this proyect. step by step Sorry for my English, Humberto Atristain V. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI questions
We are in the prossess of ordering a ISDN (PRI) line from Verizon. There are a couple of questions on the application that I don't understand. I'm still a newbie here but we're taking the big plunge. If someone could give me some direction here I would be greatly appreciative. The questions on their app that I don't understand are: 1.) The Verizon Co will outpulse 10 digits to the PBX, is this satisfactory? 2.)DNLS Digit Dialing - Prefix and Delete - (if yes, what will it be?) It's also asking for the PBX manufacturer / system model number and the PBX software / release. Should I just fill in Asterisk and the realease for these questions? Lastly, they wish to know if it is custom or national ISDN and if its national - if it is 1,2,3. How do I determine this? Thanks for any and all help. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another PRI based question
In speaking to the representative at Verizon, we came to the conclusion that DID numbers were not the correct solution; however we were told by Verizon that they could do something called assign individual numbers to the PRI. What this would in effect do is give us an additional phone number that we would like to route to a specific extension; however unlike the DID number, it would not be assigned to a specific channel. It would hunt for an available channel. What we would like to be able to do is that even though it doesn't come in on a specific channel, still be able to route it directly to a specific extension. The representative at Verizon said that we should be able to do this by having the PBX recognize the digits that come in on the line and route it to the specific extension accordingly. Is there a way to do this in asterisk? Thanks again. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgettone 100 phone Configuration
Will look into this once someone can help me with the configuration behind NAT (without NAT I have no problem) I am using v1.0.3.53 and a linksys router (the phone IP is 192.168.1.2) I´ve tried in my sip.conf with and without NAT=1. In the phone, if I set the outbound proxy to the linksys it doesn´t do anything. If I leave outbound proxy empty it registers and I can place calls but no audio either way. I have also tried setting the phone for NAT and no NAT (no STUN server). Don´t know what else to try. Can someone please help me? - Original Message - From: Greg Renouf [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 05, 2003 8:16 PM Subject: RE: [Asterisk-Users] Budgettone 100 phone Configuration I'm using v.1.0.3.58 and am experiencing that my phone crashes every time the call reaches about 45 minutes in length. Has anybody had a similar experience? -GSR -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. Besch Sent: 05 June 2003 19:03 To: [EMAIL PROTECTED] Subject: re: [Asterisk-Users] Budgettone 100 phone Configuration The updated Budgetone firmware (1.0.3.60) has indeed fixed the silent DTMF issue. By the way, Grandstream just got the silent DTMF problem fixed for me and sent me an updated revision this morning (1.0.3.60). I am just about to install it, but it may require that I debug my tftp server, which I haven't tested yet. I'll post the list as soon as I get the new version loaded. -- Stephen R. Besch, Ph.D. SachsLab 320 Cary Hall SUNY at Buffalo Buffalo, NY 14214 (716) 829-3289 x106 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip channel driver causes asterisk to crash when talking to quintum A800
Make that three of us. However, Asterisk isn't crashing, it's merely locking up with my ATA-186, but not my Cisco 7960's. My own debug included below. JT On Sat, 7 Jun 2003, shido wrote: This is the sip debug when the call went through [snip] Funnily enough I've been looking at the same problem. Will get a chance to look a bit more tomorrow. Steve SIP is acting poorly with my ATA-186 devices, and I can't narrow down exactly why. This is on code from about an hour ago, with a complete cvs update; make clean; make; make install . - Asterisk starts - various phones REGISTER - this works fine - test: calls from my 7960 to either line on my ATA-186 work fine - test: calls from my 7960 to any other destination work fine (IAX, Zap, etc.) - all of my phones are behind the same NAT, if that matters - the first call I try to place out of my ATA-186 fails (to any destination; my example uses a call to the 7960) but I see the included sip debug information on my console. No more SIP debugging information appears past that point. It is as if the ATA-186 for some reason kills Asterisk, where it did not before. - After that point, all other SIP calls from any other device fail, and looking at tethereal I see that there are no replies to new SIP REGISTER requests, either. I can type stop now or stop gracefully and the system will not stop. I have to manually killall to get asterisk to die. - I backed out to a version from June 3 21:18 and all dial modes work correctly with exactly the same /etc/asterisk/* files, so it is a change in Asterisk and not in the phones. *CLI show version Asterisk CVS-06/07/03-01:40:15 built by [EMAIL PROTECTED] on a i686 running Linux *CLI *CLI sip debug SIP Debugging Enabled Sip read: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.1.25:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=2961659159 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Expires: 300 Content-Length: 243 Content-Type: application/sdp v=0 o=2204 23257 23257 IN IP4 10.0.1.25 s=ATA186 Call c=IN IP4 10.0.1.25 t=0 0 m=audio 16386 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 11 lines Using latest request as basis request Sending to 10.0.1.25 : 5060 (non-NAT) Capabilities: us - 14, them - 268, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 *CLI *CLI *CLI show channels Channel (ContextExtensionPri ) State Appl. Data 0 active channel(s) *CLI *CLI *CLI sip show channels Peer UsernameCall ID Seq (Tx/Rx) Lag Jitter Format 10.0.1.25(None) 1852710522@ 00101/2 0ms ms 0 1 active SIP channel(s) *CLI Configuration for ATA-186 line 1: [2204] type=friend username=2204 secret=somepassword mailbox=2203 host=dynamic context=intern canreinvite=no dtmfmode=rfc2833 nat=1 For reference, here is the SIP debug for a functional call from a 7960 on the same version of Asterisk code (2203 = 7960, 2204 = ATA-186 line 1) *CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.1.15:5060 From: 2203 sip:[EMAIL PROTECTED];tag=0002b9eb0ef400c3289c4132-36211630 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Sat, 07 Jun 2003 19:19:33 GMT CSeq: 101 INVITE User-Agent: CSCO/4 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Content-Type: application/sdp Content-Length: 241 Accept: application/sdp Remote-Party-ID: 2203 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 14891 19200 IN IP4 10.0.1.15 s=SIP Call c=IN IP4 10.0.1.15 t=0 0 m=audio 23764 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 10.0.1.15 : 5060 (non-NAT) Capabilities: us - 14, them - 268, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wish-to-have in Asterisk
On Sat, 2003-06-07 at 17:33, Humberto Atristain V. wrote: Dear Pals I think all of us in this business (or most) looking at the Corn Flakes Back Side , Wish to find a NAT Friendly VOIP solution. Thats ridiculous I know, but what about this wish list (Personal Point of view) IAX is that protocol. Works out of the box across a NAT. It does this by putting the entire call on a single port and the machine behind a NAT does register updates around every 60 seconds. This keeps any half decent NAT solution happy with the port staying up and working. Upnp support for Asterisk What would this do for asterisk? I'll have to claim ignorance as to what you would accomplish with it. STUN Support for Asterisk This would be okay, as well as the proxy support. Of course now that FWD and IAXTEL are routing together, this will elliminate my needs for SIP. Small Embedded Asterisk (just to redirect IAX) SIP-IAX , h.323-IAX or something like that , on the End Point. As a Nat friendly end point How small are you thinking? What type of media are you thinking of putting this on? Looking at my current phone routing machine thats only duty is to terminate my PRI and route some calls to a channel bank in the colo rack, and send the rest down our data T1 to our office asterisk machine. On this machine It currently is taking 1gig of drive space, but of that 245M is my src directory, 119M is documentation, I have 122M in my postgres database area, and 242M in my home directory. If I remove all those spots I mentioned above, and uninstall postgres, I'd almost be able to stick this setup on a 256meg CF card. This is a default debian install with all the dev packages to compile asterisk too. So I bet I could easily build a machine that would do what you want. Why does that need to be part of the core asterisk project? the most diffuctult Integrate IAX to RFCXX SIP , or H.323 to be recognized as Standard. Go to SIP Interop forums and look at the possibility to integrate IAX to SIP( or something like that) Manufacturers include IAX as Sip, MGCP, H.323 This is more market pressure than a project goal. If there is enough people asking for that feature, then more than just the Snom folks will look into providing it. Mark and all the others, May be a good idea to start a WISH LIST for asterisk? Not for the experts in development, Just let comon Asterisk Users (Like Me) to put my bit to this interesting proyect. So if there`s a specialized developer get his Wish-List Point and start over and move forward this proyect. step by step -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI questions
On Sat, 2003-06-07 at 19:14, [EMAIL PROTECTED] wrote: We are in the prossess of ordering a ISDN (PRI) line from Verizon. There are a couple of questions on the application that I don't understand. I'm still a newbie here but we're taking the big plunge. If someone could give me some direction here I would be greatly appreciative. The questions on their app that I don't understand are: 1.) The Verizon Co will outpulse 10 digits to the PBX, is this satisfactory? See my other comment, but 10 digits is what you want. outpulse is a interesting way of putting it. Of course I went through some interesting times dealing with our telco in getting them to use terminology that was appropriate to the circuit we where ordering. 2.)DNLS Digit Dialing - Prefix and Delete - (if yes, what will it be?) not sure on this one. It's also asking for the PBX manufacturer / system model number and the PBX software / release. Should I just fill in Asterisk and the realease for these questions? May want to include your T1 card model number(T100P,T400P) too so if they want to check up on the FCC cert they can easily. Lastly, they wish to know if it is custom or national ISDN and if its national - if it is 1,2,3. How do I determine this? Thanks for any and all help. If you look in /etc/asterisk/zapata.conf you will see the options. Looks like you need national, but I don't know what version. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip channel driver causes asterisk to crash when talking to quintum A800
I experienced the exact same symptoms but didn't have the confidence to post my experience to this list because of my lack of experience with Asterisk. I restored the June 1 version from CVS and the problem went away. There's definitely a problem in code since June 1. On Sat, 7 Jun 2003, John Todd wrote: - After that point, all other SIP calls from any other device fail, and looking at tethereal I see that there are no replies to new SIP REGISTER requests, either. I can type stop now or stop gracefully and the system will not stop. I have to manually killall to get asterisk to die. - I backed out to a version from June 3 21:18 and all dial modes work correctly with exactly the same /etc/asterisk/* files, so it is a change in Asterisk and not in the phones. *CLI show version Asterisk CVS-06/07/03-01:40:15 built by [EMAIL PROTECTED] on a i686 running Linux *CLI *CLI sip debug SIP Debugging Enabled Sip read: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.1.25:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=2961659159 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Expires: 300 Content-Length: 243 Content-Type: application/sdp v=0 o=2204 23257 23257 IN IP4 10.0.1.25 s=ATA186 Call c=IN IP4 10.0.1.25 t=0 0 m=audio 16386 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 11 lines Using latest request as basis request Sending to 10.0.1.25 : 5060 (non-NAT) Capabilities: us - 14, them - 268, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 *CLI *CLI *CLI show channels Channel (ContextExtensionPri ) State Appl. Data 0 active channel(s) *CLI *CLI *CLI sip show channels Peer UsernameCall ID Seq (Tx/Rx) Lag Jitter Format 10.0.1.25(None) 1852710522@ 00101/2 0ms ms 0 1 active SIP channel(s) *CLI Configuration for ATA-186 line 1: [2204] type=friend username=2204 secret=somepassword mailbox=2203 host=dynamic context=intern canreinvite=no dtmfmode=rfc2833 nat=1 For reference, here is the SIP debug for a functional call from a 7960 on the same version of Asterisk code (2203 = 7960, 2204 = ATA-186 line 1) *CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.1.15:5060 From: 2203 sip:[EMAIL PROTECTED];tag=0002b9eb0ef400c3289c4132-36211630 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Sat, 07 Jun 2003 19:19:33 GMT CSeq: 101 INVITE User-Agent: CSCO/4 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Content-Type: application/sdp Content-Length: 241 Accept: application/sdp Remote-Party-ID: 2203 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 14891 19200 IN IP4 10.0.1.15 s=SIP Call c=IN IP4 10.0.1.15 t=0 0 m=audio 23764 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 10.0.1.15 : 5060 (non-NAT) Capabilities: us - 14, them - 268, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another PRI based question
Well I would have went with DIDs however it really increases the pricing of their plan plus we then have to split the channels up as incoming and outgoing. It gets pretty complicated. They already deliver 10 digits in from what I understand. I will inquire from them whether or not I can set my outbound callerid. Would that be setting the name or the number or both? Also its kind of too late to switch now because I have a time frame in which to have the project complete and getting PRIs in is not a quick process even from Verizon. So it is sure to be more difficult from one of the Clec's. Thanks AJ On Sat, 7 Jun 2003, Steven Critchfield wrote: On Sat, 2003-06-07 at 19:19, [EMAIL PROTECTED] wrote: In speaking to the representative at Verizon, we came to the conclusion that DID numbers were not the correct solution; however we were told by Verizon that they could do something called assign individual numbers to the PRI. What this would in effect do is give us an additional phone number that we would like to route to a specific extension; however unlike the DID number, it would not be assigned to a specific channel. It would hunt for an available channel. What we would like to be able to do is that even though it doesn't come in on a specific channel, still be able to route it directly to a specific extension. The representative at Verizon said that we should be able to do this by having the PBX recognize the digits that come in on the line and route it to the specific extension accordingly. Is there a way to do this in asterisk? Thanks again. AJ It may be time to ask for a new person to work with. You want DID numbers. You want the DIDs to be delivered as the full length number, 10 digits. This lets you put all your incoming calls into a simple context where you define extensions that direct the incoming phone number to a specific function or internal extension. While them delivering 10 digits may be overkill for DID, it allows you to get DID numbers from different exchanges without any problems. Also you may want to make sure they let you set your callerid number on outbound calls. It is helpfull for my setup since my office phones present the main number for the office. The last DID we have is what I use for my home phone, and it presents the last DIDs number so no one sees my office line as my callerid anymore. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI questions
When I look at the zapata.conf file, I see switchtype=national and its uncommented. But I'm still a little confused here cause there appears to be other options there. Is there any definitive way for me to determine which type I need? AJ On Sat, 7 Jun 2003, Steven Critchfield wrote: On Sat, 2003-06-07 at 19:14, [EMAIL PROTECTED] wrote: We are in the prossess of ordering a ISDN (PRI) line from Verizon. There are a couple of questions on the application that I don't understand. I'm still a newbie here but we're taking the big plunge. If someone could give me some direction here I would be greatly appreciative. The questions on their app that I don't understand are: 1.) The Verizon Co will outpulse 10 digits to the PBX, is this satisfactory? See my other comment, but 10 digits is what you want. outpulse is a interesting way of putting it. Of course I went through some interesting times dealing with our telco in getting them to use terminology that was appropriate to the circuit we where ordering. 2.)DNLS Digit Dialing - Prefix and Delete - (if yes, what will it be?) not sure on this one. It's also asking for the PBX manufacturer / system model number and the PBX software / release. Should I just fill in Asterisk and the realease for these questions? May want to include your T1 card model number(T100P,T400P) too so if they want to check up on the FCC cert they can easily. Lastly, they wish to know if it is custom or national ISDN and if its national - if it is 1,2,3. How do I determine this? Thanks for any and all help. If you look in /etc/asterisk/zapata.conf you will see the options. Looks like you need national, but I don't know what version. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI questions
On Sat, 2003-06-07 at 21:26, [EMAIL PROTECTED] wrote: When I look at the zapata.conf file, I see switchtype=national and its uncommented. But I'm still a little confused here cause there appears to be other options there. Is there any definitive way for me to determine which type I need? You only need what will make it work. National is fine. If your telco offered this option, jump on it. While do not know for certain, I doubt you gain much going with any other options for switch type. On Sat, 7 Jun 2003, Steven Critchfield wrote: On Sat, 2003-06-07 at 19:14, [EMAIL PROTECTED] wrote: We are in the prossess of ordering a ISDN (PRI) line from Verizon. There are a couple of questions on the application that I don't understand. I'm still a newbie here but we're taking the big plunge. If someone could give me some direction here I would be greatly appreciative. The questions on their app that I don't understand are: 1.) The Verizon Co will outpulse 10 digits to the PBX, is this satisfactory? See my other comment, but 10 digits is what you want. outpulse is a interesting way of putting it. Of course I went through some interesting times dealing with our telco in getting them to use terminology that was appropriate to the circuit we where ordering. 2.)DNLS Digit Dialing - Prefix and Delete - (if yes, what will it be?) not sure on this one. It's also asking for the PBX manufacturer / system model number and the PBX software / release. Should I just fill in Asterisk and the realease for these questions? May want to include your T1 card model number(T100P,T400P) too so if they want to check up on the FCC cert they can easily. Lastly, they wish to know if it is custom or national ISDN and if its national - if it is 1,2,3. How do I determine this? Thanks for any and all help. If you look in /etc/asterisk/zapata.conf you will see the options. Looks like you need national, but I don't know what version. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another PRI based question
On Sat, 2003-06-07 at 21:04, [EMAIL PROTECTED] wrote: Well I would have went with DIDs however it really increases the pricing of their plan plus we then have to split the channels up as incoming and outgoing. It gets pretty complicated. They already deliver 10 digits in from what I understand. I will inquire from them whether or not I can set my outbound callerid. Would that be setting the name or the number or both? Also its kind of too late to switch now because I have a time frame in which to have the project complete and getting PRIs in is not a quick process even from Verizon. So it is sure to be more difficult from one of the Clec's. Thanks I wasn't suggesting a different CLEC, just a different rep to deal with. It seems either the tariffs in your area are whacked, or you may be getting ripped. For our latest PRI install we went with Telcove, formerly Adelphia. A 20 block of DIDs costs $4 for our DIDs. We didn't have to split our lines between incoming and outgoing. This is the point of PRI, all the signalling goes on out of band to negotiate the channels. Even when we had EM lines from MCI, we had our DIDs and no splitting of the functions. These are reasons why you either need them to explain why you are being told this, or ask for a new rep that is more experienced. Or possibly see if you can't schedule a meeting with a switch tech that is used to actually configing the switch. On Sat, 7 Jun 2003, Steven Critchfield wrote: On Sat, 2003-06-07 at 19:19, [EMAIL PROTECTED] wrote: In speaking to the representative at Verizon, we came to the conclusion that DID numbers were not the correct solution; however we were told by Verizon that they could do something called assign individual numbers to the PRI. What this would in effect do is give us an additional phone number that we would like to route to a specific extension; however unlike the DID number, it would not be assigned to a specific channel. It would hunt for an available channel. What we would like to be able to do is that even though it doesn't come in on a specific channel, still be able to route it directly to a specific extension. The representative at Verizon said that we should be able to do this by having the PBX recognize the digits that come in on the line and route it to the specific extension accordingly. Is there a way to do this in asterisk? Thanks again. AJ It may be time to ask for a new person to work with. You want DID numbers. You want the DIDs to be delivered as the full length number, 10 digits. This lets you put all your incoming calls into a simple context where you define extensions that direct the incoming phone number to a specific function or internal extension. While them delivering 10 digits may be overkill for DID, it allows you to get DID numbers from different exchanges without any problems. Also you may want to make sure they let you set your callerid number on outbound calls. It is helpfull for my setup since my office phones present the main number for the office. The last DID we have is what I use for my home phone, and it presents the last DIDs number so no one sees my office line as my callerid anymore. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another PRI based question
AJ. If it's okay, I'd like to offer my 2 cents. For switch type - it really doesn't matter what you pick, as long as both ends (yours and your telco's) are configured to be same. I don't think anyone has ever noticed any advantages or disadvantages with one switch type over the other. What's more important, however, when talking to your provider is that you tell them that you want your PRI to be able to make long distance calls also. Don't assume this part or you'll only get local calls. For DID - I believe Verizon does offer a block of 25 DIDs. These DIDs will be programmed by your carrier to point to your PRI's circuit ID and it's also known as your LDN (Listed Directory Number). Once these DIDs are programmed, be sure to ask your carrier how many digits they are sending you so that you can configure your PBX to receive that many digits as well. Once your PBX is configured correctly to send and receive the right amount of digits between you and your carrier, you're now need to program these DIDs and associate them to your LENS (Line Equipment Numbers - in the NEC world), hand them out to your users. (I believe Asterisk (in fact, all PBX) can accomplished this task easily. Look at the extension.conf area. (This is just a guess on my part. I too am new to Asterisk.)) Good luck. In speaking to the representative at Verizon, we came to the conclusion that DID numbers were not the correct solution; however we were told by Verizon that they could do something called assign individual numbers to the PRI. What this would in effect do is give us an additional phone number that we would like to route to a specific extension; however unlike the DID number, it would not be assigned to a specific channel. It would hunt for an available channel. What we would like to be able to do is that even though it doesn't come in on a specific channel, still be able to route it directly to a specific extension. The representative at Verizon said that we should be able to do this by having the PBX recognize the digits that come in on the line and route it to the specific extension accordingly. Is there a way to do this in asterisk? Thanks again. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Seng [EMAIL PROTECTED] www.simplifiednetwork.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgettone 100 phone Configuration
Dan Fernandez writes: In the phone, if I set the outbound proxy to the linksys it doesn´t do anything. i have noticed this too. outbound proxy feature is broken in it. also, it doesn't do srv lookups, which would allow leaving outbound proxy empty. it looks to me that the gs guys still have ways to go before the phone is ready for prime time. -- juha ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
