Re: [asterisk-users] Asterisk Prompt
On Thu, Aug 23, 2007 at 03:13:40PM -0700, bilal ghayyad wrote: Dear Mojo; Thanks for your help. Why you said export ASTERISK_PROMPT=new prompt ? To make that a new value for the environment variable. An alternative method is: ASTERISK_PROMPT=new prompt asterisk -r There are actually some special % values there: ASTERISK_PROMPT='date(d): %d, h(hostname) : %h, H(short hostname): %H, l(load avg): %l, s(system name) %s, t(time): %t, literal: %% ' rasterisk -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
On Thu, Aug 23, 2007 at 07:16:57PM -0400, Lee Jenkins wrote: Steve Totaro wrote: David Gomillion wrote: On 8/23/07, *Ed Pastore* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, folks. I've been on the Asterisk Announce list for a while now, and it seems to me that the release versions of Asterisk are a bit bleeding-edge. They qualify as stable, but I wouldn't call them production stable since half the time a new one comes out, a fix for it comes out the next day. That's the niche that ABE is supposed to fill. I personally don't use it, though. I just test the features I plan to use, disable everything else, and seem to do OK. What version of Asterisk is current ABE (something that would get installed on a new system with no relation to other systems) based on? I stay with 1.2.12 or somewhere around there. End Of Life but seems to have a better ticker than 1.4. Thanks, Steve 1.2.12/14/17 all have seemed very stable to me so far. Both of which are anecdotial evidences. Now suppose I had a major stability issue with 1.2.14 which was solved with 1.2.18 (or 1.4.1). I would simply be dropped off that tatistics. You'l be just left with those for which 1.2 works better. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Simulating errors (Busy / Out of Order)
I'm trying to build a test suite so that I can run calls through and verify the call results. I've made a cross over cable and linked my 2 ISDN30 ports together. So now I can dial out on span 1 , and to receive the call on span 2. in the context for span 2, I have the following: snip ; #1 answer a call and play music 000XXX : ring for a random period, answer, play moh, hangup after a random period ; #2 just ring (no answer) 001XXX : ring for a random period, hangup after a random period ; #3 out of order 002XXX : Zapateller() snip ; #4 engaged 003XXX : Busy() #1 and #2 just work fine. however, with #4, I get Accepting call from '123456' to '003123' on channel 0/31, span 2 -- Executing [EMAIL PROTECTED]:1] NoOp(Zap/62-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Busy(Zap/62-1, 3) in new stack -- Zap/31-1 is proceeding passing it to SIP/5711-0834fdd0 -- Zap/31-1 is making progress passing it to SIP/5711-0834fdd0 What I was hoping was to get a busy signal on the SIP channel. I get a similar result with #3 Does anyone have an idea of what I am doing wrong here ? The dialplan for #4 is: exten = _003X.,1,NoOp() ; Engaged exten = _003X.,n,Busy(3) exten = _003X.,n,Hangup() Julian. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] recomend web interface for virual call center
Hi I Have been asked to setup a virtual call center. the server will be hosted at the ISP, with the incoming lines for the local telco .. the calls from the incomming lines will then be forwarded to individual users directly or to other call centers .. Any suggestions on web mangment interface for asterisk / call center .. and needs to be open source .. Gregory Machin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simulating errors (Busy / Out of Order)
Oh, man, why is it that when you spend hours working on something, as soon as you send a message for help, the solution presents itself ? To answer my own question, and for prosperity, see the comments inline: Sorry for the waste of bandwidth :( Julian Lyndon-Smith wrote: I'm trying to build a test suite so that I can run calls through and verify the call results. I've made a cross over cable and linked my 2 ISDN30 ports together. So now I can dial out on span 1 , and to receive the call on span 2. in the context for span 2, I have the following: snip ; #1 answer a call and play music 000XXX : ring for a random period, answer, play moh, hangup after a random period Works great ; #2 just ring (no answer) 001XXX : ring for a random period, hangup after a random period works great ; #3 out of order 002XXX : Zapateller() snip ; #4 engaged 003XXX : Busy() for these two, set the PRI_CAUSE variable before hangup I set 17 for BUSY and 27 for out of order #1 and #2 just work fine. however, with #4, I get Accepting call from '123456' to '003123' on channel 0/31, span 2 -- Executing [EMAIL PROTECTED]:1] NoOp(Zap/62-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Busy(Zap/62-1, 3) in new stack -- Zap/31-1 is proceeding passing it to SIP/5711-0834fdd0 -- Zap/31-1 is making progress passing it to SIP/5711-0834fdd0 What I was hoping was to get a busy signal on the SIP channel. I get a similar result with #3 Does anyone have an idea of what I am doing wrong here ? The dialplan for #4 is: exten = _003X.,1,NoOp() ; Engaged exten = _003X.,n,Busy(3) exten = _003X.,n,Hangup() exten = _003X.,1,NoOp() ; Engaged exten = _003X.,n,Set(PRI_CAUSE=17) exten = _003X.,n,Hangup() Julian. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email for dotr.com has been scanned by MessageLabs __ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE210P digim card PRI problem
Dear all I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back and second E1 card on Direct Telcom for outgoing for outside now i got this error when i call on avaya PRI asterisk think PRI_CPE and remote end also CPE i have configure /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-32 dchan=16 span=1,1,0,ccs,hdb3 bchan=32-46,49-62 dchan=47 in /etc/asterisk/zapata.conf switchtype=qsig context=zap-in signalling=pri_cpe group=1 channel = 1-15,17-31 group=2 channel = 32-46,49-62 is this configuration is fine or any other problem when i call to my second e1 which i connected to direct telcom i got this error call can't forward caz voice or dtmf can anyone send my runing configuration file of zaptel.conf or zapata.conf file waiting for your reply - Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem compiling Zaptel 1.4.5.1
Hi. Please help. When trying to compile Zaptel 1.4.5.1 I get the following: /build/include/linux/modversions.h -DSTANDALONE_ZAPATA -I.. -o base.o -c base.c base.c:48:29: linux/workqueue.h: No such file or directory base.c:292: warning: `vpm150m_firmware' defined but not used make[2]: *** [base.o] Error 1 make[2]: Leaving directory `/usr/src/zaptel-1.4.5.1/wctdm24xxp' make[1]: *** [wctdm24xxp/wctdm24xxp.o] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4.5.1' make: *** [all] Error 2 Can anybody help me with this? I run make distclean, configure and then make. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firefly IAX2 configuration
Dear Gordon; Thanks a lot, it is working and it was from the firewall. But what is the command that I can know all the rigestered endusers (iax2 or sip or h323)? I tried iax2 show registry but it did not give any thing? Can u help? Regards Bilal --- Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 20 Aug 2007, bilal ghayyad wrote: Dear Gordon; Thanks a lot for your email. I need one more tracing tool, how can I know the used port of the IAX on teh Asterisk and wethor the listening on that port is successully done (ready to receive on that port)? Use netstat -lnveep to list open ports and display the programs using them. About the firewall, actually the client PC and Asterisk on the same LAN (my PC is 192.168.8.2 and Asterisk is 192.168.8.4), the only possible thing is the firewall on the fedora server (Asterisk server), but I am not so friendly with fedora to know how can I check if the firewall on fedora enabled if u can help me (fedora is like redhat). I don't know fedora either, but try: iptables -n -L and it it spews forth lots and lots of lines, then there is local firewalling. You can turn all iptable firewalling off with: iptables --flush iptables --delete-chain but it will restore upon reboot (probably) Whether turning all firewalling off is a good thing or not, is up to you, but as it's on a private LAN, then I'd suggest it's probably OK. Gordon Regards Bilal Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password host=dynamic Then I ran the following: #/usr/sbin/asterisk -cvvv CLIreload But always I get a message at the firefly that an error occured while trying to connect to the network. What else I have to do? Have you checked your firewall? Is it letting UDP data through to the asterisk box on port 4569? By the way: what is the command that I can type it to do tracing on the user [iax2user1] or to do traces on any registeration attempts from the clients? iax2 debug will generate lots of output for you... Last thing, if I am outside the console (in unix mode), is there any command from unix I can type it to know if asterisk is running or not? ps ax | grep asterisk is crude, but visual. Asterisk stores it's PID in /var/run/asterisk.pid, so you could then read that, and check to see if the process with that PID is actually running asterisk. ie. see if /proc/number existis, and if-so, see if it's actually asterisk by reading /proc/number/cmdline or just see if you can connect to it with the rasterisk command ... Gordon Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error in loading libunicall.so module while running asterisk command
On 8/23/07, [EMAIL PROTECTED] Hi, I am using debian 4.0 with version 2.6.18-4-686 I have downloaded the required files form site asterisk-1.2.24.tar.gz libmfcr2-0.0.3-1.4.tar.bz2 libsupertone-0.0.2.tar.gz libunicall-0.0.3-1.4.tar.bz2 spandsp-20060903.tar.gz I downloaded and installed the files in the follwing sequence spandsp libsupertone libunicall libmfcr2-0.0.3 is giving a lot of definition error I converted .src.rpm file of libmfcr2 to .deb file and installed it. the copying the chn_unicall.c and channels_Makefile.patch to channels subdirectory of asterisk-1.2.24 but when I run ,asterisk -vvgc' on command line it gives following error message -- loader.c: 326 __load_resource:libunicall.so.0cannot open shared object filer loader.c:555load_modulesloading module chan_unicall.so failed but libunicall.so is present. Can you tell me how to trobleshoot it. Thanka and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE210P digim card PRI problem
Hi, you have to correct your etc/zaptel.conf as follows span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 then try Regards, Vidura == I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back and second E1 card on Direct Telcom for outgoing for outside now i got this error when i call on avaya PRI asterisk think PRI_CPE and remote end also CPE i have configure /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-32 dchan=16 span=1,1,0,ccs,hdb3 bchan=32-46,49-62 dchan=47 in /etc/asterisk/zapata.conf switchtype=qsig context=zap-in signalling=pri_cpe group=1 channel = 1-15,17-31 group=2 channel = 32-46,49-62 is this configuration is fine or any other problem when i call to my second e1 which i connected to direct telcom i got this error call can't forward caz voice or dtmf can anyone send my runing configuration file of zaptel.conf or zapata.conf file waiting for your reply === ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file
Dear All, Am happy to say that I've successfully been able to register a SIP user from a soft phone terminal via LDAP. The biggest hurdle that I had to overcome was the LDAP-Asterisk schema. The schema example given in the astirectory installation document is incomplete. Here's are a few pointers in this regard: The attributes have to be defined in the following way. Also tab spaces should be avoided. dn: cn=schema changetype: modify add: attributetypes attributeTypes: ( 1.3.6.1.4.1.23935.5.4.1 NAME 'astUsername' DESC '' SUP name EQUALITY caseIgnoreMatch SUBSTR caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) NAME should be the same as objectIdentifier DESC should be the description of the attribute EQUALITY is the rule to use when doing a search/compare for an attribute value. SUBSTR is the rule to use when doing a substring search (*foo*) SYNTAX is the syntax (i.e., type) of the attribute. We should probably stick to syntaxes: 1.3.6.1.4.1.1466.115.121.1.15 - directoryString (UTF-8 string) 1.3.6.1.4.1.1466.115.121.1.26 - IA5String (ASCII String) 1.3.6.1.4.1.1466.115.121.1.27 - integer (Integer value) The object class has to be always defined as AUXILLARY and never ABSTRACT. dn: cn=schema changetype: modify add: objectclasses objectClasses: ( 1.3.6.1.4.1.23935.5.5.1 NAME 'astSipGeneric' DESC '' SUP top AUXILIARY MUST ( astContext ) MAY ( astSecret $ astPermit $ astDeny $ astMd5Secret $ astDtmfmode $ astCanreinvite $ astNat $ astCallgroup $ astPickupgroup $ astAllow $ astDisallow $ astInsecure $ astTrustrpid $ astProgressinband $ astPromiscredir $ astRegseconds $ astname $ astLanguage ) ) Best Regards Abhishek On 8/16/07, Anthony Francis [EMAIL PROTECTED] wrote: You will need to extend your schema to include all of the attributes that can be used in sip.conf plus the extra ones that allow realtime to store connection information. Please refer to the realtime info at voipinfo.org to get a feel for what your schema should look like. Anthony Abhishek M S wrote: Dear all, May I first introduce myself. I'm a student of HAW Hamburg University currently working for my professor on a VOIP project. We have a Debian Linux system (server) on which Asterisk 1.2.6 has been successfully installed and running. Also the asterisk SIP server has been connected to the PSTN so users could make calls externally. We use Xlite softphone to make calls between users in the network. Currently there are very few users and I have been able to register them in the in *sip.conf *file and declare extensions in the *extensions.conf *file. Now there is a requirement to assign extensions to all students in the university(over thousand) whose credentials and information is stored in the Novel based LDAP database. Moving along I've managed to successfully install astirectory which is a real time database driver that allows to fetch configuration data from LDAP directories. Have also installed the LDAPget module that can lookup data in the LDAP directory. I'm looking for SIP attributes on LDAP or an LDAP schema that would facilitate astirectory or LDAPget to retrieve the username, telephone number and password from the LDAP database to register the soft phone user. I'd be extremely grateful for any help or suggestion in this connection. Thanks in advance, Abhishek ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MYSQL problem and configuration
Greetings everyone I've set up a call recording system on debian 4.0 with asterisk and mysql db for handling user information (accessible over the net for users). My asterisk is running on one machine and the mysql on another. The connection is over lan. Now I have a problem and a question. My problem is: When mysql_real_connect doesn't get connection to the mysql server asterisk pretty much freezes and doesn't let any info go in or out, even the CLI freezes. I've seen a bug report on this but no solution(?) Although this might be a bug with asterisk when I have the connection set to a hosted mysql server (company that hosts our website during the testing period) the connection works fine so I've come to conclusion that the problem might be the mysql and/or debian configuration I have. SO... My question is: How should I configure mysql (and debian box) for asterisk connections? Best regards Jari-Pekka Lehtinen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE210P digim card PRI problem
You need to look at pri_cpe vs pri_net. PaulH On Fri, 2007-08-24 at 01:27 -0700, satish patel wrote: Dear all I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back and second E1 card on Direct Telcom for outgoing for outside now i got this error when i call on avaya PRI asterisk think PRI_CPE and remote end also CPE i have configure /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-32 dchan=16 span=1,1,0,ccs,hdb3 bchan=32-46,49-62 dchan=47 in /etc/asterisk/zapata.conf switchtype=qsig context=zap-in signalling=pri_cpe group=1 channel = 1-15,17-31 group=2 channel = 32-46,49-62 is this configuration is fine or any other problem when i call to my second e1 which i connected to direct telcom i got this error call can't forward caz voice or dtmf can anyone send my runing configuration file of zaptel.conf or zapata.conf file waiting for your reply __ Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL problem and configuration
Hello,I am new to asterisk but i have vbeen scriptinh PHP SQL and webLanguages
for a long time.I can Give you a solution but using php
AGI:extensions.con- AGI(connect.agi);/var/lib/asterisk/agi-bin/connect.agi
:#!/usr/bin/php
-q?phpset_time_limit(0);ob_implicit_flush();error_reporting(0);//Initialisation
des entrée-sortiefunction init() {#create file handles if
neededif(!defined('STDIN')){
define('STDIN',fopen('php://stdin','r'));}if(!defined('STDOUT')){
define('STDOUT',fopen('php://stdout','w'));}if(!defined('STDERR')){
define('STDERR',fopen('php://stderr','w'));}#retrieve all AGI variables from
Asteriskwhile(!feof(STDIN)){$temp=trim(fgets(STDIN,4096));
if(($temp==)||($temp==\n)){break;}$s=split(:,$temp);
$name=str_replace(agi_,,$s[0]);$agi[$name]=trim($s[1]);}return
$agi;}function checkresult($res){trim($res);
if(preg_match('/^200/',$res)){
if(!preg_match('/result=(-?\d+)/',$res,$matches)){
fwrite(STDERR,FAIL ($res)\n);fflush(STDERR);return0;
} else {fwrite(STDERR,PASS (.$matches[1].)\n);
fflush(STDERR);return $matches[1];}} else {
fwrite(STDERR,FAIL (unexpected result '$res')\n);fflush(STDERR);
return -1;}}$agivar = init();$hostname= '';
$database= 'x';
$username= 'x';
$password= '';
$dbprotect = mysql_pconnect($hostname, $username, $password) or
trigger_error(mysql_error(),E_USER_ERROR);
mysql_select_db($database, $dbprotect);$result = mysql_query(SELECT * FROM
user_table WHERE user_age12);while($entry = mysql_fetch_array($result))
{fwrite(STERR, Name : $entry['name'], Age: $entry['age']
\n);fflush(STDOUT);$result =
trim(fgets(STDIN,4096));checkresult($result);}?It will return things on the
asterisk CLI You can adapt this example for youI don't know if it help but
it shows a way to do...Kheraud From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Date: Fri, 24 Aug 2007 12:56:55 +0300
Subject: [asterisk-users] MYSQL problem and configuration Greetings
everyone I've set up a call recording system on debian 4.0 with asterisk and
mysql db for handling user information (accessible over the net for users).
My asterisk is running on one machine and the mysql on another. The
connection is over lan. Now I have a problem and a question. My problem
is: When mysql_real_connect doesn't get connection to the mysql server
asterisk pretty much freezes and doesn't let any info go in or out, even the
CLI freezes. I've seen a bug report on this but no solution(?) Although this
might be a bug with asterisk when I have the connection set to a hosted mysql
server (company that hosts our website during the testing period) the
connection works fine so I've come to conclusion that the problem might be the
mysql and/or debian configuration I have. SO... My question is: How should
I configure mysql (and debian box) for asterisk connections? Best regards
Jari-Pekka Lehtinen ___
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Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file
Please see the official tracker in the Digium buglist: http://bugs.digium.com/view.php?id=5768 Here are the schemas we did for OpenLDAP: http://bugs.digium.com/file_download.php?file_id=14842type=bug http://bugs.digium.com/file_download.php?file_id=14841type=bug Also, for Novell eDirectory, see: http://forge.voicerd.org/frs/?group_id=7release_id=17 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem compiling Zaptel 1.4.5.1
On Fri, Aug 24, 2007 at 08:29:55AM +, Jan du Toit wrote: Hi. Please help. When trying to compile Zaptel 1.4.5.1 I get the following: /build/include/linux/modversions.h -DSTANDALONE_ZAPATA -I.. -o base.o -c base.c base.c:48:29: linux/workqueue.h: No such file or directory base.c:292: warning: `vpm150m_firmware' defined but not used make[2]: *** [base.o] Error 1 make[2]: Leaving directory `/usr/src/zaptel-1.4.5.1/wctdm24xxp' make[1]: *** [wctdm24xxp/wctdm24xxp.o] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4.5.1' make: *** [all] Error 2 Can anybody help me with this? I run make distclean, configure and then make. What kernel version do you use, exactly? What linux distribution? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech Rec on Voicemail
Stephen Bosch wrote: Ryan M. Colbert wrote: I’ve had requests to processes incoming voicemails with voice recognition routine and add the output text to the body of the email message from * with the attached .wav file. Has anyone implemented this type of feature and willing to share some notes? I seem to recall that Lt. Cmdr Jordi Laforge did some stuff like this not too long ago. I get requests like this all the time -- but the technology is very far from being there. -Stephen- Your best bet at this point is to have an assistant or some other 3rd party transcribe the VM for you. I know of several companies that do this for recordings in various industries. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Fwd: Re: issues with caller ID , remote-party-id
Hello ppl,
Sorry to re-post it, but kinda these issues are getting on my nerves.
I tried Set(CALLERID(num)=7329) on 1.2.12, which works fine, but not on
1.4.4.
The problem :
1. I receive call from caller 'AAA' on my number, 'BBB' which is on my
Asterisk box.
2. I have to redirect the call to some other number, say, my cell num -
'CCC'.
3. My PSTN provider wants the calling(From) number as 'BBB', which is
fair enough, because that number has been assigned to me by this provider.
4. I have been able to achieve this(using Set(CALLERD(num)='BBB'), on
1.2.12, but not on 1.4.4. I know, be default, From will be set to BBB,
but still
5. But, more importantly, I need to pass the original caller number too
to the destination, i.e. to my cell fone - CCC, which shows up on my
cell fone as the caller id.
6. I presume, this can be achieved using Remote-Party-ID.
7. If I set sendrpid=yes in sip.conf, the stuff sent in Remote-Party-ID
also is CCC, but I want it to be AAA (actual caller).
8. So, I commented out sendrpid, and manually added Remote-Party-ID using:
SIPAddHeader(Remote-Party-ID: MEUSER AAA\;privacy=off\;screen=no)
9. As I am experimenting, I don't really have PSTN connectivity yet, but
I have come to know, that most devices, like CISCO 7960, give preference
to Remote-Party-ID over the From number to show as Caller ID. So, I have
CCC configured on a CISCO SIP phone. But the caller id is still 'BBB'.
10. And at the end of all this, I am very close to smash my asterisk
box, cisco phones with a sledgehammer.
Any bright ideas anywhere???
Help appreciated.
Thanks..
- Ben.
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---BeginMessage---
Also,
. if I use Remote-party-id header, can it be different from the 'From' URI?
. If yes, how do you achieve this in Asterisk?
. What(From or Remote-party-id) is used by clients to show as the CLI of
the caller?
if I am not mistaken, Remote-party-id is for network elements to confirm
identities of end subscribers.
All corrections and suggestions welcome.
- Ben
Benjamin Jacob wrote:
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong, Set(CALLERID(all) above supposed to change the
display name as above(Ben Jacob) and change the From URI to [EMAIL PROTECTED]
As of now, only the _display name_ is being replaced, but not the name.
I tried CALLERID(num) as well CALLERID(number), to the same effect(only
display name being set to number).
Anyone facing similar problems?
Thanks in advance.
- Ben
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whom they are addressed. Any unauthorised distribution or copying is strictly
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sender by reply email and then destroy the message. Opinions, conclusions and
other information in this message that do not relate to official business of
Mascon shall be understood to be neither given nor endorsed by Mascon. Any
information contained in this email, when addressed to Mascon clients is
subject to the terms and conditions in governing client contract.
Whilst Mascon takes steps to prevent the transmission of viruses via e-mail,
we can not guarantee that any email or attachment is free from computer
viruses and you are strongly advised to undertake your own anti-virus
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Re: [asterisk-users] channel not hungup (zombie?) so call limit not reset to zero
here is the sip debug for the channel. Before reading the sip debug i want ot tell you that user is using Telco Systems AC-211 v4.50.27 adapter. sip sdebug shows that asterisk is trying to send the initial invite but there is no response from the user (after registration, user dies, no single response). So maybe there is some kind of network issue (NAT) or there is something wrong with the Telco Systems adapter. The stuck channels are still there: IP crunch 260bca1e59e 00102/0 unkn No Init: INVITE IP crunch 350d1a6e2b1 00102/0 unkn No Init: INVITE and core show channels show 0 active calls *CLI core show channels Channel Location State Application(Data) 0 active channels 0 active calls SIP DEBUG Audio is at 64.182.161.2 port 10678 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 72.73.66.175:50069: INVITE sip:[EMAIL PROTECTED]:9060 SIP/2.0 Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport From: adf xyz sip:[EMAIL PROTECTED]:9060;tag=as22ac8da7 To: sip:[EMAIL PROTECTED]:9060 Contact: sip:[EMAIL PROTECTED]:9060 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 24 Aug 2007 09:54:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 334 v=0 o=root 28236 28236 IN IP4 64.182.161.2 s=session c=IN IP4 64.182.161.2 t=0 0 m=audio 10678 RTP/AVP 0 8 18 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #1 (NAT) to 72.73.66.175:50069: INVITE sip:[EMAIL PROTECTED]:9060 SIP/2.0 Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport From: adf xyz sip:[EMAIL PROTECTED]:9060;tag=as22ac8da7 To: sip:[EMAIL PROTECTED]:9060 Contact: sip:[EMAIL PROTECTED]:9060 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 24 Aug 2007 09:54:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 334 v=0 o=root 28236 28236 IN IP4 64.182.161.2 s=session c=IN IP4 64.182.161.2 t=0 0 m=audio 10678 RTP/AVP 0 8 18 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (NAT) to 72.73.66.175:50069: INVITE sip:[EMAIL PROTECTED]:9060 SIP/2.0 Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport From: adf xyz sip:[EMAIL PROTECTED]:9060;tag=as22ac8da7 To: sip:[EMAIL PROTECTED]:9060 Contact: sip:[EMAIL PROTECTED]:9060 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 24 Aug 2007 09:54:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 334 v=0 o=root 28236 28236 IN IP4 64.182.161.2 s=session c=IN IP4 64.182.161.2 t=0 0 m=audio 10678 RTP/AVP 0 8 18 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (NAT) to 72.73.66.175:50069: INVITE sip:[EMAIL PROTECTED]:9060 SIP/2.0 Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport From: adf xyz sip:[EMAIL PROTECTED]:9060;tag=as22ac8da7 To: sip:[EMAIL PROTECTED]:9060 Contact: sip:[EMAIL PROTECTED]:9060 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 24 Aug 2007 09:54:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 334 v=0 o=root 28236 28236 IN IP4 64.182.161.2 s=session c=IN IP4 64.182.161.2 t=0 0 m=audio 10678 RTP/AVP 0 8 18 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #4 (NAT) to 72.73.66.175:50069: INVITE sip:[EMAIL PROTECTED]:9060 SIP/2.0 Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport From: adf xyz sip:[EMAIL PROTECTED]:9060;tag=as22ac8da7 To: sip:[EMAIL PROTECTED]:9060 Contact: sip:[EMAIL PROTECTED]:9060 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 24 Aug 2007 09:54:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 334 v=0 o=root 28236 28236 IN IP4
Re: [asterisk-users] Stable-Stable Asterisk
Tzafrir Cohen wrote: On Thu, Aug 23, 2007 at 07:16:57PM -0400, Lee Jenkins wrote: Steve Totaro wrote: David Gomillion wrote: On 8/23/07, *Ed Pastore* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, folks. I've been on the Asterisk Announce list for a while now, and it seems to me that the release versions of Asterisk are a bit bleeding-edge. They qualify as stable, but I wouldn't call them production stable since half the time a new one comes out, a fix for it comes out the next day. That's the niche that ABE is supposed to fill. I personally don't use it, though. I just test the features I plan to use, disable everything else, and seem to do OK. What version of Asterisk is current ABE (something that would get installed on a new system with no relation to other systems) based on? I stay with 1.2.12 or somewhere around there. End Of Life but seems to have a better ticker than 1.4. Thanks, Steve 1.2.12/14/17 all have seemed very stable to me so far. Both of which are anecdotial evidences. Now suppose I had a major stability issue with 1.2.14 which was solved with 1.2.18 (or 1.4.1). I would simply be dropped off that tatistics. You'l be just left with those for which 1.2 works better. You lost me with that last statement All I know is that dropping 200 calls is bad and it happens less with certain versions. The tatistics or statistics are determined in my mind by the number of higher ups frantically calling and barging in demanding WHAT HAPPENED, AND HOW CAN WE PREVENT THIS FROM EVER HAPPENING AGAIN?. THIS WAS NEVER HAPPENED WITH OUR (INSERT ANY SWITCH HERE), WE JUST LOST $26,000, ASTERISK SUCKS! Of course it OCCASIONALLY happened with the old switch, and it took half an hour to an hour to reboot that switch. The proof is the pudding as they say. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] recomend web interface for virual call center
Gregory Machin wrote: Hi I Have been asked to setup a virtual call center. the server will be hosted at the ISP, with the incoming lines for the local telco .. the calls from the incomming lines will then be forwarded to individual users directly or to other call centers .. Any suggestions on web mangment interface for asterisk / call center .. and needs to be open source .. Gregory Machin How involved is the logic in the routing decisions (skills based, time based, DNIS based, ANI based, (dynamic or static) metrics based of agent or call center performance? Aheeva makes a killer product but far from open source. Other than that, you may have to roll your own. If you are simply just sending calls to any available agent, you could probably get by with most of the GUIs out there (with slight mods). Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firefly IAX2 configuration
On Fri, 24 Aug 2007, bilal ghayyad wrote: Dear Gordon; Thanks a lot, it is working and it was from the firewall. But what is the command that I can know all the rigestered endusers (iax2 or sip or h323)? I tried iax2 show registry but it did not give any thing? Can u help? sip show peers iax2 show peers Gordon Regards Bilal --- Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 20 Aug 2007, bilal ghayyad wrote: Dear Gordon; Thanks a lot for your email. I need one more tracing tool, how can I know the used port of the IAX on teh Asterisk and wethor the listening on that port is successully done (ready to receive on that port)? Use netstat -lnveep to list open ports and display the programs using them. About the firewall, actually the client PC and Asterisk on the same LAN (my PC is 192.168.8.2 and Asterisk is 192.168.8.4), the only possible thing is the firewall on the fedora server (Asterisk server), but I am not so friendly with fedora to know how can I check if the firewall on fedora enabled if u can help me (fedora is like redhat). I don't know fedora either, but try: iptables -n -L and it it spews forth lots and lots of lines, then there is local firewalling. You can turn all iptable firewalling off with: iptables --flush iptables --delete-chain but it will restore upon reboot (probably) Whether turning all firewalling off is a good thing or not, is up to you, but as it's on a private LAN, then I'd suggest it's probably OK. Gordon Regards Bilal Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password host=dynamic Then I ran the following: #/usr/sbin/asterisk -cvvv CLIreload But always I get a message at the firefly that an error occured while trying to connect to the network. What else I have to do? Have you checked your firewall? Is it letting UDP data through to the asterisk box on port 4569? By the way: what is the command that I can type it to do tracing on the user [iax2user1] or to do traces on any registeration attempts from the clients? iax2 debug will generate lots of output for you... Last thing, if I am outside the console (in unix mode), is there any command from unix I can type it to know if asterisk is running or not? ps ax | grep asterisk is crude, but visual. Asterisk stores it's PID in /var/run/asterisk.pid, so you could then read that, and check to see if the process with that PID is actually running asterisk. ie. see if /proc/number existis, and if-so, see if it's actually asterisk by reading /proc/number/cmdline or just see if you can connect to it with the rasterisk command ... Gordon Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE210P digim card PRI problem
Thnk for reply can you tell me one thing what is the meaning of this line span=2,0,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 why did u use 2,0,0 ? one more thing can you send me your config file i want to see more option send me your /etc/zaptel.cong and /etc/asterisk/zapata.conf file Regards satish patel Vidura Senadeera [EMAIL PROTECTED] wrote: Hi, you have to correct your etc/zaptel.conf as follows span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 then try Regards, Vidura == I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back and second E1 card on Direct Telcom for outgoing for outside now i got this error when i call on avaya PRI asterisk think PRI_CPE and remote end also CPE i have configure /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-32 dchan=16 span=1,1,0,ccs,hdb3 bchan=32-46,49-62 dchan=47 in /etc/asterisk/zapata.conf switchtype=qsig context=zap-in signalling=pri_cpe group=1 channel = 1-15,17-31 group=2 channel = 32-46,49-62 is this configuration is fine or any other problem when i call to my second e1 which i connected to direct telcom i got this error call can't forward caz voice or dtmf can anyone send my runing configuration file of zaptel.conf or zapata.conf file waiting for your reply === ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Shape Yahoo! in your own image. Join our Network Research Panel today!___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE120P digium card PRI_CPE error
Dear all I got one more error my asterisk E1 card connected with avaya E1 card [avaya]---E1-[asterisk] i got this 2 error what is start asteris on consol mode asterisk -c [Jul 27 09:51:29] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:30] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:31] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:32] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. 2-- second error [Jul 27 09:51:32] WARNING[737] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! - Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE210P digim card PRI problem
Search the wiki, Google it, you will feel better than being spoon fed. It is all about timing on the E1. Who is providing the timing and who is taking it. There is plenty of info on www.voip-info.org that will explain this much more than a few responses on the user's list. Thanks, Steve satish patel wrote: Thnk for reply can you tell me one thing what is the meaning of this line span=2,0,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 why did u use 2,0,0 ? one more thing can you send me your config file i want to see more option send me your /etc/zaptel.cong and /etc/asterisk/zapata.conf file Regards satish patel */Vidura Senadeera [EMAIL PROTECTED]/* wrote: Hi, you have to correct your etc/zaptel.conf as follows span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 then try Regards, Vidura == I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back and second E1 card on Direct Telcom for outgoing for outside now i got this error when i call on avaya PRI asterisk think PRI_CPE and remote end also CPE i have configure /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-32 dchan=16 span=1,1,0,ccs,hdb3 bchan=32-46,49-62 dchan=47 in /etc/asterisk/zapata.conf switchtype=qsig context=zap-in signalling=pri_cpe group=1 channel = 1-15,17-31 group=2 channel = 32-46,49-62 is this configuration is fine or any other problem when i call to my second e1 which i connected to direct telcom i got this error call can't forward caz voice or dtmf can anyone send my runing configuration file of zaptel.conf or zapata.conf file waiting for your reply ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE120P digium card PRI_CPE error
Set it to pri_net instead of pri_cpe. IF you start getting error messages that We think we are NET and they think they are NET, then your carrier or the Avaya has the line in Loopback mode. satish patel wrote: Dear all I got one more error my asterisk E1 card connected with avaya E1 card [avaya]---E1-[asterisk] i got this 2 error what is start asteris on consol mode asterisk -c [Jul 27 09:51:29] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:30] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:31] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:32] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. 2-- second error [Jul 27 09:51:32] WARNING[737] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it posible for an incoming to ring to Polycom and cell at the same time?
Andrew Kohlsmith wrote: On Thursday 23 August 2007 11:22:23 pm Stephen Bosch wrote: dial(SIP/polycom-on-my-deskLocal/5551212,15,tr) Will this work even if the Local is pointing to a Zap channel? As far as I know, this only works with SIP or IAX outgoing. I'm not sure where you are getting that assumption from, as I have been Dialing Zap/fooZap/bar, SIP/fooSIP/bar, IAX/fooIAX/bar and combinations of all three for the past several years. That's not what was in your example. Your example is a mix of Zap and SIP. Zap channels answer immediately, so if you do Dial() to multiple technologies, the Zap() channel will always answer first. I don't think that's what the original poster was looking for. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it posible for an incoming to ring to Polycom and cell at the same time?
Stephen Bosch wrote: That's not what was in your example. Your example is a mix of Zap and SIP. Zap channels answer immediately, so if you do Dial() to multiple technologies, the Zap() channel will always answer first. This is not quite accurate; Zap channels that are analog FXO ports answer immediately. FXS channels don't (they don't answer until the other end does), and all digital Zap channels have proper answer supervision. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech Rec on Voicemail
Just for starter, look at CallWave, and Jott. -E ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bluetooth questions
I see that the term now is chan_mobile to use a bluetooth to cellphone trunk. (what is in a name? :) ) What I want to know is: Is there any restriction on the bluetooth chipset for the server? Can I use the dongle for PAN and chan_mobile at the same time? Can I use the dongle for headset (a separate extension) and a cellphone at the same time? Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
On Thu, 2007-08-23 at 18:42 -0300, [EMAIL PROTECTED] wrote: 1- I've tried running fxotune 2- I've tried turning off all un-necessary hardware in the BIOS 3- I've tried on a different PCI slot. 4- I've tried these suggestions: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting 5- How I check if it the clicking and popping correlates to hard drive activity ? 6- I've not tried installing this board in another PC to test my FXOs 7- I've an MSI motherboard and AMD athlon 64 x2 Dual core processor 8- I've Turning off echotraining. Digium offers installation support on their hardware cards, so if you continue to experience problems, Digium support should be able to help you track down the cause of the problem. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech Rec on Voicemail
On 8/24/07, EdPimentl [EMAIL PROTECTED] wrote: Just for starter, look at CallWave, and Jott. -E They seem to be commercial :( A quick search in google revealed a page with some compilation of opensource STT engines. http://www.faqs.org/docs/Linux-HOWTO/Speech-Recognition-HOWTO.html Making them process voicemail wav files isn't trivial but shouldn't be quite hard. If you have some progress on this, give us know. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
Hello Jared, On 8/24/07, Jared Smith [EMAIL PROTECTED] wrote: Digium offers installation support on their hardware cards, so if you continue to experience problems, Digium support should be able to help you track down the cause of the problem. I also have the same issue on my TDM400P card but I am not in the US so I don't know how I can call to your number, 877-546-8963 for free. Thanks in advance. GNUbie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it posible for an incoming to ring to Polycom and cell at the same time?
Stephen Bosch wrote: Anthony Francis wrote: dial(SIP/polycom-on-my-deskLocal/5551212,15,tr) Will this work even if the Local is pointing to a Zap channel? As far as I know, this only works with SIP or IAX outgoing. -Stephen- I use local because It dials the call from default using your established dialing patterns, if the number is an outside number my system passes it out the zap trunk like a good little switch and I don't have to worry where the number is. -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
snip Until 1.4 improves, I'm staying with 1.2 /snip Ditto ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Keeping queue counters after restarting
Hello, Every queue has some status counters (completed, abandoned, hold time...) that are very useful for statistics. The problem is that those counters are reset every time Asterisk restarts. Is there a way to keep those counters, maybe in astdb? Also, is there a way to reset the counters through a cli command? Thanks. -- MARLON DUTRA Propus GnuPG ID: 0x3E2060AC pgp.mit.edu http://www.propus.com.br/ http://hackers.propus.com.br/~marlon/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote: Tzafrir Cohen wrote: On Thu, Aug 23, 2007 at 07:16:57PM -0400, Lee Jenkins wrote: Steve Totaro wrote: I stay with 1.2.12 or somewhere around there. End Of Life but seems to have a better ticker than 1.4. Thanks, Steve 1.2.12/14/17 all have seemed very stable to me so far. Both of which are anecdotial evidences. Now suppose I had a major stability issue with 1.2.14 which was solved with 1.2.18 (or 1.4.1). I would simply be dropped off those statistics. You'l be just left with those for which 1.2 works better. You lost me with that last statement Suppose you are a reader of a specific mailing list. Someone asked which is better: 1.2 or 1.4. Naturally the sample size you get is very small: only a handful of the large body of Asterisk users actually naswered it. I was windering if it is also skewed in any way. In fact, I pointed out one wat it can be. For instance, following the same logic, I'd say that Asterisk 1.0 is more stable than 1.2, as people have been using it for much longer in production. Nobody has been using an 1.2 PBX in production for more than, say, three years and 1.0 has been used for longer than that. So 1.0 must be more stable. Admins still using it mut probably swear by it. But most people (at least those who have had problem, including stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and now swear (by?) 1.2 or 1.4. Cheers, -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
On Fri, Aug 24, 2007 at 10:09:19PM +0800, GNUbie wrote: Hello Jared, On 8/24/07, Jared Smith [EMAIL PROTECTED] wrote: Digium offers installation support on their hardware cards, so if you continue to experience problems, Digium support should be able to help you track down the cause of the problem. I also have the same issue on my TDM400P card but I am not in the US so I don't know how I can call to your number, 877-546-8963 for free. One option: get an account in sipphone.com. A voip-ionly account is free, and still allows you calling US toll-free numbers and various others. http://www.sipphone.com/numbers/ -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't create audio conversation between softphones through Asterisk
Hello, I have two user machines, each with a jain-sip-applet-phone installed on it. I use the following process to try to make a call: 1. Register each phone with the Asterisk server (working). 2. Add a contact in each phone which is the other user. (Get a 489 Bad Event SIP error shown below in red) [EMAIL PROTECTED] has been added to your contacts. null send request: SUBSCRIBE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE From: sip:[EMAIL PROTECTED];tag=8505 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585 Max-Forwards: 2 Contact: sip:[EMAIL PROTECTED]:8386;transport=udp Content-Length: 0 message from=192.168.1.251:8386 to=192.168.1.10:5060 time=1187721756281 isSender=true transactionId=z9hg4bk361290cad5885dbc4a03b5951cc85585 callId=[EMAIL PROTECTED] firstLine=SUBSCRIBE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 debugLine=0 ![CDATA[SUBSCRIBE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE From: sip:[EMAIL PROTECTED];tag=8505 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585 Max-Forwards: 2 Contact: sip:[EMAIL PROTECTED]:8386;transport=udp Content-Length: 0 ]] /message message from=192.168.1.10:5060 to=192.168.1.251:8386 time=1187721756281 isSender=false statusMessage=normal processing transactionId=z9hg4bk361290cad5885dbc4a03b5951cc85585 firstLine=SIP/2.0 489 Bad Event callId=[EMAIL PROTECTED] debugLine=0 ![CDATA[SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585;received=192.168.1.251 From: sip:[EMAIL PROTECTED];tag=8505 To: sip:[EMAIL PROTECTED];tag=as2cf724e9 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY Content-Length: 0 3. Try to call that contact to create an audio conversation.(Get a 488 Not Acceptable Here SIP error shown below in blue) Get chat session: [EMAIL PROTECTED] Chat Session added: [EMAIL PROTECTED]:gov.nist.applet.phone.ua.gui.ChatFrame[frame0,0,0,750x430,invalid,hidden,layout=java.awt.BorderLayout,title=In conversation with [EMAIL PROTECTED],resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,23,104x1,invalid,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true] 5 4 3 0 send request: INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: sip:[EMAIL PROTECTED];tag=2085 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2 Max-Forwards: 2 Contact: sip:[EMAIL PROTECTED]:8386;transport=udp Content-Type: application/sdp Content-Length: 114 v=0 o=201 908031 909400 IN IP4 192.168.1.251 s=- c=IN IPV4 192.168.1.251 t=0 0 m=audio 2448 RTP/AVP 5 4 3 0 message from=192.168.1.251:8386 to=192.168.1.10:5060 time=1187721758593 isSender=true transactionId=z9hg4bk583c7ea7c1a8da8576583356f821c9c2 callId=[EMAIL PROTECTED] firstLine=INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 debugLine=0 ![CDATA[INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: sip:[EMAIL PROTECTED];tag=2085 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2 Max-Forwards: 2 Contact: sip:[EMAIL PROTECTED]:8386;transport=udp Content-Type: application/sdp Content-Length: 114 ]] /message message from=192.168.1.10:5060 to=192.168.1.251:8386 time=1187721758609 isSender=false statusMessage=normal processing transactionId=z9hg4bk583c7ea7c1a8da8576583356f821c9c2 firstLine=SIP/2.0 488 Not acceptable here callId=[EMAIL PROTECTED] debugLine=0 ![CDATA[SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2;received=192.168.1.251 From: sip:[EMAIL PROTECTED];tag=2085 To: sip:[EMAIL PROTECTED];tag=as2f851644 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY Content-Length: 0 Has anyone ever tried using these Jain-sip-applet-phones and got them to work? I have read up on these errors, and it looks like the 489 error doesn't like the SUBSCRIBE request, while the 488 error doesn't seem to accept the INVITE request made. I am not sure if this is a problem with Asterisk, incompatibility between Asterisk and the phones, or just the phones. Any thoughts that may help me resolve these issues would be greatly appreciated. Thanks very much, Denis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] DTFM not recognise
Hello,Maybe I don't understand what DTMF in ASCII means but I can't make my record stop using this syntax in a PHP agi script :fwrite(STDOUT, RECORD FILE /var/lib/asterisk/ENR/jeanpaul wav '#' 15000 BEEP s=3000\n);The php syntax isn't a problem because I really start recording, I have a beep, the record can't long more than 15sec and after 3sec of silence my record stop. Btu if I press # it doesn't stop the record.It i probably a problem with my '#' syntax.Can you explain me how to fix that problem. (I have the same problem with SAY NUMBer and the escape DTMF...Thanks again for your useful helpKheraud _ Besoin d'un e-mail ? Créez gratuitement un compte Windows Live Hotmail, plus sûr, plus simple et plus complet ! http://www.windowslive.fr/hotmail/default.asp___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Keeping queue counters after restarting
On 8/24/07, Marlon Dutra [EMAIL PROTECTED] wrote: Hello, Every queue has some status counters (completed, abandoned, hold time...) that are very useful for statistics. The problem is that those counters are reset every time Asterisk restarts. Is there a way to keep those counters, maybe in astdb? Also, is there a way to reset the counters through a cli command? Nope. Plus i personally don't think that they are much of use. You should be processing queue_log or CDR to obtain more complete picture. But if you feel you really need it you can post a feature request in bugs.digium.com. It would be called something like persistent queue status - in analogy with persistent agents and persistent queues. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Keeping queue counters after restarting
On 8/24/07, Marlon Dutra [EMAIL PROTECTED] wrote: Every queue has some status counters (completed, abandoned, hold time...) that are very useful for statistics. The problem is that those counters are reset every time Asterisk restarts. Is there a way to keep those counters, maybe in astdb? Also, is there a way to reset the counters through a cli command? Not sure about restarts, but trunk keeps them through reloads. How often are you restarting? From http://svn.digium.com/view/asterisk/trunk/CHANGES?revision=79638view=markup : Queue changes - * Added keepstats option to queues.conf which will keep queue statistics during a reload. I don't think there's a command to reset the counters - would be a good (and relatively simple I think) patch to offer up before 1.6 gets closer to a release. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] xPL and Asterisk?
On Fri, 2007-08-24 at 03:44 -0500, [EMAIL PROTECTED] wrote: Message: 20 Date: Thu, 23 Aug 2007 23:13:55 -0500 From: Jay Milk [EMAIL PROTECTED] Subject: Re: [asterisk-users] xPL and Asterisk? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Matthew Rubenstein wrote: I tried asking in another thread this week, but I'm not sure people saw the actual subject of the question. Does anyone know where to find documentation of xPL, the home automation interface? Specifically for integrating it with Asterisk. xPL is part of Trixbox, so it's being used, but where is some expertise for using it without Trixbox? http://www.google.com/search?q=xpl+home+automation 1st and 3rd results. I actually mentioned the explicit Google search URL in my previous message to the list. But I also mentioned that I prefer the list's experience in actual use of xPL with Asterisk. I'm looking for specific xPL/Asterisk docs that Asterisk people have tested. The community is a source of best practices, which is what I'm looking for. Like insight into whether to use the xPLhub for Linux that's available, or whether there's a different way to go. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
Tzafrir Cohen wrote: On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote: stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and now swear (by?) 1.2 or 1.4. My decision based on what I've been reading in the bug tracker and people commenting on how they've had to roll back to 1.2 to regain a stable system. We are not having issues with our 1.2.x installs, but I've been 'encouraged' by the development team to upgrade to 1.4. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNOW Web GUI
Is the web GUI for AsteriskNOW able to be loaded on an existing server(that was installed from ubuntu-server and asterisk loaded from source)? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't create audio conversation between softphonesthrough Asterisk
This is the full log that I get after my trial run: Aug 24 14:15:51 VERBOSE[3710] logger.c: -- Registered SIP '202' at 192.168.1.250 port 9810 expires 120 Aug 24 14:15:52 VERBOSE[3710] logger.c: -- Registered SIP '201' at 192.168.1.251 port 8529 expires 120 Aug 24 14:15:55 NOTICE[3710] chan_sip.c: Peer '202' is now UNREACHABLE! Last qualify: 0 Aug 24 14:15:56 NOTICE[3710] chan_sip.c: Peer '201' is now UNREACHABLE! Last qualify: 0 Aug 24 14:16:07 DEBUG[3710] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Aug 24 14:16:07 DEBUG[3710] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Aug 24 14:16:10 DEBUG[3710] chan_sip.c: Setting NAT on RTP to 0 Aug 24 14:16:10 WARNING[3710] chan_sip.c: Invalid host in c= line, 'IN IPV4 192.168.1.251' Aug 24 14:16:10 DEBUG[3710] chan_sip.c: SIP message could not be handled, bad request: b475318241b3dca93128681e6f079093 192.168.1.251 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Friday, August 24, 2007 10:41 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Can't create audio conversation between softphonesthrough Asterisk Hello, I have two user machines, each with a jain-sip-applet-phone installed on it. I use the following process to try to make a call: 1. Register each phone with the Asterisk server (working). 2. Add a contact in each phone which is the other user. (Get a 489 Bad Event SIP error shown below in red) [EMAIL PROTECTED] has been added to your contacts. null send request: SUBSCRIBE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE From: sip:[EMAIL PROTECTED];tag=8505 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585 Max-Forwards: 2 Contact: sip:[EMAIL PROTECTED]:8386;transport=udp Content-Length: 0 message from=192.168.1.251:8386 to=192.168.1.10:5060 time=1187721756281 isSender=true transactionId=z9hg4bk361290cad5885dbc4a03b5951cc85585 callId=[EMAIL PROTECTED] firstLine=SUBSCRIBE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 debugLine=0 ![CDATA[SUBSCRIBE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE From: sip:[EMAIL PROTECTED];tag=8505 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585 Max-Forwards: 2 Contact: sip:[EMAIL PROTECTED]:8386;transport=udp Content-Length: 0 ]] /message message from=192.168.1.10:5060 to=192.168.1.251:8386 time=1187721756281 isSender=false statusMessage=normal processing transactionId=z9hg4bk361290cad5885dbc4a03b5951cc85585 firstLine=SIP/2.0 489 Bad Event callId=[EMAIL PROTECTED] debugLine=0 ![CDATA[SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585;received=192.168.1.251 From: sip:[EMAIL PROTECTED];tag=8505 To: sip:[EMAIL PROTECTED];tag=as2cf724e9 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY Content-Length: 0 3. Try to call that contact to create an audio conversation.(Get a 488 Not Acceptable Here SIP error shown below in blue) Get chat session: [EMAIL PROTECTED] Chat Session added: [EMAIL PROTECTED]:gov.nist.applet.phone.ua.gui.ChatFrame[frame0,0,0,750x430,invalid,hidden,layout=java.awt.BorderLayout,title=In conversation with [EMAIL PROTECTED],resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,23,104x1,invalid,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true] 5 4 3 0 send request: INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: sip:[EMAIL PROTECTED];tag=2085 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2 Max-Forwards: 2 Contact: sip:[EMAIL PROTECTED]:8386;transport=udp Content-Type: application/sdp Content-Length: 114 v=0 o=201 908031 909400 IN IP4 192.168.1.251 s=- c=IN IPV4 192.168.1.251 t=0 0 m=audio 2448 RTP/AVP 5 4 3 0 message from=192.168.1.251:8386 to=192.168.1.10:5060 time=1187721758593 isSender=true transactionId=z9hg4bk583c7ea7c1a8da8576583356f821c9c2 callId=[EMAIL PROTECTED] firstLine=INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 debugLine=0 ![CDATA[INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: sip:[EMAIL PROTECTED];tag=2085 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2 Max-Forwards: 2 Contact: sip:[EMAIL PROTECTED]:8386;transport=udp Content-Type: application/sdp Content-Length: 114 ]] /message message from=192.168.1.10:5060 to=192.168.1.251:8386 time=1187721758609 isSender=false statusMessage=normal processing
Re: [asterisk-users] AsteriskNOW Web GUI
You can do it from svn server , I think there is a page in the wiki Best, yann _ De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jeremy Mann Envoyé : vendredi 24 août 2007 17:30 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] AsteriskNOW Web GUI Is the web GUI for AsteriskNOW able to be loaded on an existing server(that was installed from ubuntu-server and asterisk loaded from source)? _ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by http://www.mailscanner.info/ MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] channel not hungup (zombie?) so call limit not reset to zero
finally found a solution for stuck channels. Im sharing this for anyone who may face this problem in future. It seems a bug as i can repeat this behaviour both ways as many times as i want. i will start an issue for this on digium bug site but first i have to test it on the latest version of asterisk. Asterisk loses binding with peers somehow if qualify=no is set in sip.conf. when asterisk loses binding it does not know that the peer is now UNREACHABLE becoz it does not monitor it all the time due to qualify=no. So when a call comes for that (UNREACHABLE) user asterisk tries to send sip packet for that user which the user does not recieve. Asterisk has no problem with that, it only throws call to voicemail or whatever is defined in dialplan. But after hangup channel get stuck. If we set qualify=yes, then asterisk keeps track of users whether they are REACHABLE or not. So if any user is UNREACHABLE it knows about that user and does not bother to send sip packets to that user anymore. This way channel is not even initialesed if sip invite is recieved for that channel (and goes directly to voicemail) and uninitailised channels cannot get stuck. On 8/24/07, Rizwan Hisham [EMAIL PROTECTED] wrote: here is the sip debug for the channel. Before reading the sip debug i want ot tell you that user is using Telco Systems AC-211 v4.50.27 adapter. sip sdebug shows that asterisk is trying to send the initial invite but there is no response from the user (after registration, user dies, no single response). So maybe there is some kind of network issue (NAT) or there is something wrong with the Telco Systems adapter. The stuck channels are still there: IP crunch 260bca1e59e 00102/0 unkn No Init: INVITE IP crunch 350d1a6e2b1 00102/0 unkn No Init: INVITE and core show channels show 0 active calls *CLI core show channels Channel Location State Application(Data) 0 active channels 0 active calls SIP DEBUG Audio is at 64.182.161.2 port 10678 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 72.73.66.175:50069: INVITE sip:[EMAIL PROTECTED] :9060 SIP/2.0 Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport From: adf xyz sip:[EMAIL PROTECTED]:9060;tag=as22ac8da7 To: sip:[EMAIL PROTECTED] :9060 Contact: sip:[EMAIL PROTECTED]:9060 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 24 Aug 2007 09:54:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 334 v=0 o=root 28236 28236 IN IP4 64.182.161.2 s=session c=IN IP4 64.182.161.2 t=0 0 m=audio 10678 RTP/AVP 0 8 18 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #1 (NAT) to 72.73.66.175:50069: INVITE sip:[EMAIL PROTECTED]:9060 SIP/2.0 Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport From: adf xyz sip:[EMAIL PROTECTED]:9060;tag=as22ac8da7 To: sip:[EMAIL PROTECTED]:9060 Contact: sip:[EMAIL PROTECTED]:9060 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 24 Aug 2007 09:54:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 334 v=0 o=root 28236 28236 IN IP4 64.182.161.2 s=session c=IN IP4 64.182.161.2 t=0 0 m=audio 10678 RTP/AVP 0 8 18 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (NAT) to 72.73.66.175:50069: INVITE sip:[EMAIL PROTECTED]:9060 SIP/2.0 Via: SIP/2.0/UDP 64.182.161.2:9060 ;branch=z9hG4bK473eaef8;rport From: adf xyz sip:[EMAIL PROTECTED]:9060;tag=as22ac8da7 To: sip:[EMAIL PROTECTED]:9060 Contact: sip:[EMAIL PROTECTED]:9060 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 24 Aug 2007 09:54:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 334 v=0 o=root 28236 28236 IN IP4 64.182.161.2 s=session c=IN IP4 64.182.161.2 t=0 0 m=audio 10678 RTP/AVP 0 8 18 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (NAT) to 72.73.66.175:50069: INVITE sip:[EMAIL PROTECTED]:9060 SIP/2.0
Re: [asterisk-users] Stable-Stable Asterisk
While it is not exactly running a huge system, I have had one 1.4 system running in a small office of 10 phones since June with no problems and another small system for about a month with no problems. I have also had a larger system (80+ phones, DUNDi and IAX trunking to 11 sites) running 1.4 for a over a month. That system has had stability issues from time to time with the IAX, I account most of the issues I have had to the changes being made and the fact that 90% of the systems it interacts with are 1.2 versions. I know there are bugs in 1.4, as are there bugs in 1.2 and likely even in 1.0. I did not move to 1.4 to avoid bugs or fix anything, but to use certain features to accomplish goals that the client had for the system. I think Tzafrir is right: --- Suppose you are a reader of a specific mailing list. Someone asked which is better: 1.2 or 1.4. Naturally the sample size you get is very small: only a handful of the large body of Asterisk users actually naswered it. So I am answering as someone using 1.4. Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
Doug Lytle wrote: Tzafrir Cohen wrote: On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote: stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and now swear (by?) 1.2 or 1.4. My decision based on what I've been reading in the bug tracker and people commenting on how they've had to roll back to 1.2 to regain a stable system. We are not having issues with our 1.2.x installs, but I've been 'encouraged' by the development team to upgrade to 1.4. I'll just chime in for those who are thinking of moving to 1.4 and do end up having issues... don't just turn around and go back to 1.2 immediately. File a bug report with all the needed information so things can get fixed. As a development team we can't test every single scenario possible with Asterisk, we depend on the users to tell us if there are problems and tell us how to reproduce them. Asterisk only gets better thanks to the users out there. If you file a bug report keep on top of it... if more information is needed, provide it. I've had a few bugs where the reporter dropped off the radar and I had to end up trying every possible configuration combination to find the bug and fix it, taking away time that I could have spent on other issues. I'm going to end this email with a question myself... how many people have Asterisk on a development/staging server before deployment, test, and isolate the issues they may have in their specific scenario? -- Joshua Colp Software Developer Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Prompt
(sent to the list because it's pertinent) Bilal, anything can be a unix variable, it doesn't matter what it's called. #!/bin/bash export THREE=3 export FIVE=5 ((EIGHT=$THREE+$FIVE)) echo $EIGHT THREE, FIVE, and EIGHT are all just variables, like in programming. It just so happens that if you create one called ASTERISK_PROMPT, asterisk will use it. You would probably put this in your startup script, for example /etc/rc.d/rc.local or /etc/rc.local down near the bottom you could put export ASTERISK_PROMPT=new prompt and when you run asterisk, it will use this new prompt in the CLI. bilal ghayyad wrote: Dear Mojo; Where we configure this UNIX environment that can let us able to write the command export ASTERISK_PROMPT (for example, how can we know that if we typed export CISCO_PROMPT then it will not work we CISCO_PROMPT is not UNIX variable)? Regards Bilal --- Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: You seemed to be unclear about unix variables: The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? the command: export ASTERISK_PROMPT= will let you change it, like you asked for example, if your asterisk server is named pbx, you could get your default prompt back with: export ASTERISK_PROMPT=pbx*CLI or you could play around: export ASTERISK_PROMPT=Enter your command to the Asterisk CLI Sorry if this is unclear! Mojo bilal ghayyad wrote: Dear Mojo; Thanks for your help. Why you said export ASTERISK_PROMPT=new prompt ? Regards Bilal I'm not sure what features/variables you can use, or where to find information about that, but what this basically means is you can change your CLI prompt by this: export ASTERISK_PROMPT=new prompt then, what you access the CLI, instead of: hostname*CLI you get new prompt Moj bilal ghayyad wrote: Hi List; I read the following sentence: The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable In the following link: http://www.voip-info.org/wiki/index.php page=Asterisk+CLI+prompt The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? Also where I can find information about it? Regards Bilal Ghayad Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center. http://autos.yahoo.com/green_center/ Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech Rec on Voicemail
I'd be interesting in pooling resources for this. We've seen the success of Vonage's Visual Voicemail and would like to emulate a similar solution. Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mitcheloc Sent: Friday, August 24, 2007 1:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Speech Rec on Voicemail Nuance offers an SDK to do something similar, I think they say you can only expect between 45-60% accuracy using it though. Total cost is about $6K to $8K for one server license. If there are enough people interested in pooling money I'd be willing to help set up a system to process voicemails and provide the Nuance converted transcript. However, I figure the low accuracy would be the biggest turn off from using Nuance. On 8/23/07, Stephen Bosch [EMAIL PROTECTED] wrote: Ryan M. Colbert wrote: I've had requests to processes incoming voicemails with voice recognition routine and add the output text to the body of the email message from * with the attached .wav file. Has anyone implemented this type of feature and willing to share some notes? I seem to recall that Lt. Cmdr Jordi Laforge did some stuff like this not too long ago. I get requests like this all the time -- but the technology is very far from being there. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as a softswitch
What is a good softswitch that is also open source rather than asterisk? On 8/24/07, James Jones [EMAIL PROTECTED] wrote: Yes you could, but asterisk was designed to be a PBX. I would not use it as soft switch due its limitations. It really depends on how much traffic you are going to be passing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 24 August 2007 1:11 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk as a softswitch Mark Quitoriano wrote: Can i use asterisk as a softswitch? This thread has been discussed over and over. Search the archives, there are more thoughts and opinions there than you probably have time or desire to read. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.4/969 - Release Date: 23/08/2007 4:04 p.m. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.4/969 - Release Date: 23/08/2007 4:04 p.m. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/registerr=19441 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech Rec on Voicemail
On Friday 24 August 2007 12:37, Ryan M. Colbert wrote: I'd be interesting in pooling resources for this. We've seen the success of Vonage's Visual Voicemail and would like to emulate a similar solution. Please define success, I have a vonage account, and the transcription is very poor at best. Ron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
On 8/24/07, Joshua Colp [EMAIL PROTECTED] wrote: I'm going to end this email with a question myself... how many people have Asterisk on a development/staging server before deployment, test, and isolate the issues they may have in their specific scenario? I do, but many of the problems I have experienced (see #10199 for an example) don't manifest under anything but production loads. In that particular case, I couldn't find a way to replicate the levels of traffic and the nuances of agent pickup / ignore / hangup / etc. in my lab. My current load test consists of a lab box generating about 50-75 concurrent calls to an ITSP that terminate on another * conencted to PRI. But what you do with a call when it hits your box can make a difference. I had a load test that just walked through my IVRs pressing random keys for about 5 minutes. I could load 4 PRI full of calls to that context and the box would be fine. The second I added queueing (so that there was SIP signalling out to agent softphones), I'd get a kernel panic. The agent didn't even have to pick up the phone - just making it ring was enough. Let me ask a question myself: what kind of regression test does * undergo before release, and what level of traffic gets put through stuff like app_queue? I assume it's not real-world scale, else these hard to pin down concurrency issues we're seeing would have been caught in test. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
Joshua Colp wrote: I'm going to end this email with a question myself... how many people have Asterisk on a development/staging server before deployment, test, and isolate the issues they may have in their specific scenario? I do, but with limited resources for testing (2 polycoms, no end users and no PRI) it's difficult for find issues until after a system is put into production. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
James FitzGibbon wrote: On 8/24/07, *Joshua Colp* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm going to end this email with a question myself... how many people have Asterisk on a development/staging server before deployment, test, and isolate the issues they may have in their specific scenario? I do, but many of the problems I have experienced (see #10199 for an example) don't manifest under anything but production loads. In that particular case, I couldn't find a way to replicate the levels of traffic and the nuances of agent pickup / ignore / hangup / etc. in my lab. My current load test consists of a lab box generating about 50-75 concurrent calls to an ITSP that terminate on another * conencted to PRI. But what you do with a call when it hits your box can make a difference. I had a load test that just walked through my IVRs pressing random keys for about 5 minutes. I could load 4 PRI full of calls to that context and the box would be fine. The second I added queueing (so that there was SIP signalling out to agent softphones), I'd get a kernel panic. The agent didn't even have to pick up the phone - just making it ring was enough. Let me ask a question myself: what kind of regression test does * undergo before release, and what level of traffic gets put through stuff like app_queue? I assume it's not real-world scale, else these hard to pin down concurrency issues we're seeing would have been caught in test. Open source Asterisk has no real regression testing before release. As we work on things we test and a few of us use it at home (like myself). The time and resources involved in regression testing Asterisk are just huge, which is why it is limited to business edition. As bugs are found in business edition though they are fixed in the open source version as well. That is why I asked about the testing people do ahead of time, I was curious how many people do it. -- Joshua Colp Software Developer Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
James FitzGibbon wrote: Let me ask a question myself: what kind of regression test does * undergo before release, and what level of traffic gets put through stuff like app_queue? I assume it's not real-world scale, else these hard to pin down concurrency issues we're seeing would have been caught in test. Let *me* ask a question. :) What level of heavy regression testing would you *expect* of an open source development team? We really do try very hard to test all of our changes. We have community members that work very hard to help test out the more invasive changes. Furthermore, we have a lot of people run the code from the release branches directly so that regressions are caught quicker, and hopefully before they make it in to a release. At Digium, we have a department dedicated to doing testing of our products, including Asterisk. Every bug that is found as a part of this testing gets fixed in the open source branches as well. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
On Fri, 2007-08-24 at 22:09 +0800, GNUbie wrote: I also have the same issue on my TDM400P card but I am not in the US so I don't know how I can call to your number, 877-546-8963 for free. You can call Digium at +1-256-428-6000 (which obviously wouldn't be a free call, but at least you can get through). Also, you can call over VoIP by dialing IAX2/[EMAIL PROTECTED]/6000. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNOW Web GUI
svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow thegui; cd thegui; sh configure; make sudo make install ; clear ; echo 'completed' -bk Yann JOUANIN wrote: You can do it from svn server , I think there is a page in the wiki Best, yann *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Jeremy Mann *Envoyé :* vendredi 24 août 2007 17:30 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* [asterisk-users] AsteriskNOW Web GUI Is the web GUI for AsteriskNOW able to be loaded on an existing server(that was installed from ubuntu-server and asterisk loaded from source)? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by *MailScanner* http://www.mailscanner.info/, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tuning a ZyWALL for Asterisk
I understand this question is over-broad, but hopefully you can have patience with a newbie and toss me a bone... I am in the testing stage of deploying Asterisk. I have successfully configured it to work behind the NAT of my ZyXEL ZyWALL 35 firewall. However, I think there is a lot of tuning I can do to get better reliability, bandwidth management, and maybe QoS from the firewall. I have some clues as to how to do some of this, but both telephony and routing are not strong points for me (I mostly work on systems, servers, and LANs). Is there any sort of reference material that will guide me in setting up my ZyWALL for VoIP? I don't see much help from ZyXEL, and I only see scattered posts around the net, but I know a lot of people are using ZyWALLs with Asterisk. If there isn't a reference, then can anyone chime in with some particulars on what you've done? Any hints would be greatly appreciated. Thanks! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
On 8/24/07, Joshua Colp [EMAIL PROTECTED] wrote: Doug Lytle wrote: Tzafrir Cohen wrote: On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote: stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and now swear (by?) 1.2 or 1.4. I'm going to end this email with a question myself... how many people have Asterisk on a development/staging server before deployment, test, and isolate the issues they may have in their specific scenario? First, I must say that the current Asterisk 1.2 tree IS more stable than the last 1.0 release under heavy load, it also has many more features which make the choice between 1.0 and 1.2 a no-brainer. As for 1.4 It is FAR better today than when it was first released as 1.4.0. A lot of the critical bugs have been fixed, and the more complex ones are left that are harder to reproduce, but those are getting fixed as well. I applaude Digium for putting more resources into bug fixing, and it has made a noticable difference in the 1.4 tree. With all of that said, I do have a testing setup that allows me to run tests at high loads on Asterisk, but not all scenarios can be checked in a testing setup. I ran a mid-volume test on 1.4.10 and it worked without crashing. I wanted to test a new feature in 1.4 so I put the server into production. It worked fine for a few hours under small load, but once the load increased there were several issues(mostly relating to stuck locks I am guessing) and the server would crash every few hours and also have some weird Manager API issues. So after a few days I rolled the server back to 1.2.X and all was well again. Running the tests again later at a higher call volume and on servers with more horsepower revealed the same crashes and other issues as I noticed in production. I would like to donate my services to Digium to install a performance testing setup, like I have, at Digium headquarters. They just need to supply the two servers, two quad T1 cards and 4 crossover T1 cables, and I will setup VICIDIAL and show anyone there how to run it in performance testing mode. It is great at exposing performance problems and it generates pretty graphs to boot. Someone at Digium can contact me off-list if they are interested. MATT--- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
Joshua Colp wrote: Doug Lytle wrote: Tzafrir Cohen wrote: On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote: stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and now swear (by?) 1.2 or 1.4. My decision based on what I've been reading in the bug tracker and people commenting on how they've had to roll back to 1.2 to regain a stable system. We are not having issues with our 1.2.x installs, but I've been 'encouraged' by the development team to upgrade to 1.4. I'll just chime in for those who are thinking of moving to 1.4 and do end up having issues... don't just turn around and go back to 1.2 immediately. File a bug report with all the needed information so things can get fixed. As a development team we can't test every single scenario possible with Asterisk, we depend on the users to tell us if there are problems and tell us how to reproduce them. Asterisk only gets better thanks to the users out there. If you file a bug report keep on top of it... if more information is needed, provide it. I've had a few bugs where the reporter dropped off the radar and I had to end up trying every possible configuration combination to find the bug and fix it, taking away time that I could have spent on other issues. I'm going to end this email with a question myself... how many people have Asterisk on a development/staging server before deployment, test, and isolate the issues they may have in their specific scenario? I always do. A dev system with a a few calls here and there. Checkout the cool new features, syntaxes, and nuances. Here is a better question. How can I setup a Dev system to handle 15,000 calls a day using nine queues, 200 agents, nine servers, seven quad port T1 Sangoma boards connected to a Adtran DS3 MUX? Now add specific scenarios such as AGI (of whatever form) manager connections controlling your CRM, screenpop, sales processes, and reporting. You cannot scale that up on a dev system. Now imagine if the system is down you lose $26,000/hr. You know what, I am going back to 1.2.x as quick as possible, then I will try to assemble some data and open a bug, but good luck with that. This exact same scenario played out on 1.2. It worked flawlessly until the manager flaked out under REAL load. Thanks, Steve Dev systems are great, so are small PBXs for companies that get a few dozen calls a day. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
I stay with 1.2.12 or somewhere around there. End Of Life but seems to have a better ticker than 1.4. Thanks, Steve 1.2.12/14/17 all have seemed very stable to me so far. Both of which are anecdotial evidences. Now suppose I had a major stability issue with 1.2.14 which was solved with 1.2.18 (or 1.4.1). I would simply be dropped off those statistics. You'l be just left with those for which 1.2 works better. You lost me with that last statement Suppose you are a reader of a specific mailing list. Someone asked which is better: 1.2 or 1.4. Naturally the sample size you get is very small: only a handful of the large body of Asterisk users actually naswered it. I was windering if it is also skewed in any way. In fact, I pointed out one wat it can be. For instance, following the same logic, I'd say that Asterisk 1.0 is more stable than 1.2, as people have been using it for much longer in production. Nobody has been using an 1.2 PBX in production for more than, say, three years and 1.0 has been used for longer than that. So 1.0 must be more stable. Admins still using it mut probably swear by it. But most people (at least those who have had problem, including stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and now swear (by?) 1.2 or 1.4. Cheers, I would make the comparison of a fruit such as a peach as it ages. 1.0 is over-ripe or rotten/forgotten and thrown away. Besides, we all know that 1.0 was just a marketing ploy to legitimize Asterisk. What serious company is going to install 1.BETA2 or .90? Maybe a nonessential piece of software but not something as mission critical as a PBX. 1.2 is sitting at the fresh fruit market. It is a nice peach color, soft, sweet and juicy, most of the bad peaches have been discarded such as worm and bug infestations. It has been aged perfectly. 1.4 is still a bunch of peaches on the tree. It is far from ripe and is still very green. It is prime lunch for bugs, worms, and other infestation which will not get sorted out until they get ready for the market. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
Still true on CentOS 5. You can only RAID partitions unless you do the LVM thing. What are the disadvantages compared to being able to RAID the whole disk? Maybe for monitoring it's just more to deal with but does it make a RAID 1 any less reliable? -Original Message- From: Zane C.B. [mailto:[EMAIL PROTECTED] Sent: Thursday, August 23, 2007 9:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk On Wed, 22 Aug 2007 12:37:26 -0600 Stephen Bosch [EMAIL PROTECTED] wrote: Zane C.B. wrote: 1: Software RAID on Linux is way less than impressive. Plus last a I checked Linux can't handle mirroring a entire disk. Last I looked at it around a year ago you were limited to only mirroring partitions, which is a joke from a administrative standpoint. How is this any different in FreeBSD? Could you explain to me how else you are going to mirror an entire disk in software when your boot partition is on the disk? The raid info is done the same as on other decent system, it is stored at the in the last sector of the provider. making a mirrored freebsd system is like this... 1: install freebsd 2: dd if=current drive of=2nd drive for mirror 3: gmirror label some name 2nd drive 4: mount 2nd drive and edit fstab to boot using /dev/gmirror/whatever 5: boot from 2nd drive 6: gmirror insert name original drive /me loves GEOM, the goddess of all disk subsystems or whatever. http://www.freebsd.org/cgi/man.cgi?query=gmirrorapropos=0sektion=0manpath =FreeBSD+6.2-RELEASEformat=html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem compiling Zaptel 1.4.5.1
It compiles fine for me but I can't change the soft EC. It always compiles with MG1 no matter what I select in zconfig.h. Downgraded back to 1.4.4 and it works fine again. -Original Message- From: Jan du Toit [mailto:[EMAIL PROTECTED] Sent: Friday, August 24, 2007 1:30 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problem compiling Zaptel 1.4.5.1 Hi. Please help. When trying to compile Zaptel 1.4.5.1 I get the following: /build/include/linux/modversions.h -DSTANDALONE_ZAPATA -I.. -o base.o -c base.c base.c:48:29: linux/workqueue.h: No such file or directory base.c:292: warning: `vpm150m_firmware' defined but not used make[2]: *** [base.o] Error 1 make[2]: Leaving directory `/usr/src/zaptel-1.4.5.1/wctdm24xxp' make[1]: *** [wctdm24xxp/wctdm24xxp.o] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4.5.1' make: *** [all] Error 2 Can anybody help me with this? I run make distclean, configure and then make. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
We have been running 1.4 since July 06 (it was trunk then), and have upgraded often with the 1.4 branch (Currently on SVN-branch-1.4-r77571). We have 100+ extensions (SIP) and 30 ISDN channels. We often have 50+ agents available for outbound calls and queues (20+ queues). We are making / receiving approx 5000+ calls per day. We use jabber and odbc heavily (updating / reading / Creating) as well as using odbc for cdr records. All calls are recorded (monitor at the moment). We use SMS inbound and outbound. This is on a dell 2850 with 2gb ram (top - 21:31:11 up 246 days). Asterisk has System uptime: 3 weeks, 4 days, 7 hours, 57 minutes, 44 seconds Whilst nowhere near the levels of some other people, for our purposes, 1.4 is working very very well for us, and the development guys have our gratitude and respect. It's a damn fine piece of work that has saved my company a lot of money in the 2 years we've been using asterisk. Thanks Guys ! Julian. Bruce Reeves wrote: While it is not exactly running a huge system, I have had one 1.4 system running in a small office of 10 phones since June with no problems and another small system for about a month with no problems. I have also had a larger system (80+ phones, DUNDi and IAX trunking to 11 sites) running 1.4 for a over a month. That system has had stability issues from time to time with the IAX, I account most of the issues I have had to the changes being made and the fact that 90% of the systems it interacts with are 1.2 versions. I know there are bugs in 1.4, as are there bugs in 1.2 and likely even in 1.0. I did not move to 1.4 to avoid bugs or fix anything, but to use certain features to accomplish goals that the client had for the system. I think Tzafrir is right: --- Suppose you are a reader of a specific mailing list. Someone asked which is better: 1.2 or 1.4. Naturally the sample size you get is very small: only a handful of the large body of Asterisk users actually naswered it. So I am answering as someone using 1.4. Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email for dotr.com has been scanned by MessageLabs __ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Keeping queue counters after restarting
I'm pretty sure that a command to reset the counters was added soon after this patch. Julian. James FitzGibbon wrote: On 8/24/07, Marlon Dutra [EMAIL PROTECTED] wrote: Every queue has some status counters (completed, abandoned, hold time...) that are very useful for statistics. The problem is that those counters are reset every time Asterisk restarts. Is there a way to keep those counters, maybe in astdb? Also, is there a way to reset the counters through a cli command? Not sure about restarts, but trunk keeps them through reloads. How often are you restarting? From http://svn.digium.com/view/asterisk/trunk/CHANGES?revision=79638view=markup : Queue changes - * Added keepstats option to queues.conf which will keep queue statistics during a reload. I don't think there's a command to reset the counters - would be a good (and relatively simple I think) patch to offer up before 1.6 gets closer to a release. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
Russell Bryant wrote: James FitzGibbon wrote: Let me ask a question myself: what kind of regression test does * undergo before release, and what level of traffic gets put through stuff like app_queue? I assume it's not real-world scale, else these hard to pin down concurrency issues we're seeing would have been caught in test. Let *me* ask a question. :) What level of heavy regression testing would you *expect* of an open source development team? We really do try very hard to test all of our changes. We have community members that work very hard to help test out the more invasive changes. Furthermore, we have a lot of people run the code from the release branches directly so that regressions are caught quicker, and hopefully before they make it in to a release. At Digium, we have a department dedicated to doing testing of our products, including Asterisk. Every bug that is found as a part of this testing gets fixed in the open source branches as well. I don't think anyone is arguing that you guys are not trying your hardest. The point is that 1.4.x is not stable enough for production. Start thinking in terms of the traditional telephony world and not in terms of the software world. Traditional PBXs and switches have years and years of testing and bugs are very minimal (there are many I have found in various older Avayas and Toshibas but still usually something like you cannot delete something after you create it, nothing that takes out a system). Software is pushed out as fast as possible with known bugs. I just don't really see that in the traditional telephony world. Almost every company relies on their phone system more than even email. Email could be down for a few hours, a day, even a week and business would get done. Not so with a phone system. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
Doug Lytle wrote: Tzafrir Cohen wrote: On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote: stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and now swear (by?) 1.2 or 1.4. My decision based on what I've been reading in the bug tracker and people commenting on how they've had to roll back to 1.2 to regain a stable system. We are not having issues with our 1.2.x installs, but I've been 'encouraged' by the development team to upgrade to 1.4. Doug I would tell your development team that this is a mission critical system and not a desktop PC. Unless you must have a feature only available in 1.4, leave your mission critical systems alone. Patch when necessary, and upgrade when needed. My 2 cents Alex ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
Steve Totaro wrote: 1.0 is over-ripe or rotten/forgotten and thrown away. Besides, we all know that 1.0 was just a marketing ploy to legitimize Asterisk. What serious company is going to install 1.BETA2 or .90? Maybe a nonessential piece of software but not something as mission critical as a PBX. A marketing ploy? Are you serious? Asterisk 1.0 was the point in time where there were more than enough people using Asterisk in production to justify maintaining feature frozen releases. It meant that those people no longer had to use the development code on their production machines and could start focusing on a release that wasn't a moving target. I maintained Asterisk 1.0 for about a year and a half through my own personal *volunteer* efforts. It was absolutely not a Marketing ploy, as I didn't get paid for those nights that I stayed up all night reviewing the bug fixes that Mark had been making in the development code so that I could backport them. 1.2 is sitting at the fresh fruit market. It is a nice peach color, soft, sweet and juicy, most of the bad peaches have been discarded such as worm and bug infestations. It has been aged perfectly. 1.4 is still a bunch of peaches on the tree. It is far from ripe and is still very green. It is prime lunch for bugs, worms, and other infestation which will not get sorted out until they get ready for the market. Mmm ... peaches. Anyway, this transition from Asterisk 1.2 to 1.4 has been a very interesting learning experience. We will definitely benefit from all of this when it comes around to the next time that we do a major release. I have really come to understand the different expectations of stability that people have of their phone system versus other software. The transition between 1.0 and 1.2 was a different animal. There were some really major features added between 1.0 and 1.2 that a lot of people decided it was worth running the development tree to get as opposed to waiting for 1.2 to be released. The realtime configuration architecture is one example. So, 1.2 got a lot more production use before it was actually released, and there wasn't quite the same flood of people all starting to use it at once like we have had with 1.4. Now, with Asterisk 1.4, I think we have a couple of challenges. We have the fact that there are now a *lot* more installations out there than there was at the time of 1.2 being released as the project is growing rapidly. Also, I think a lot more people have been content with the feature set of 1.2 and haven't been as eager to upgrade. So, 1.4 didn't receive as much production use before it got officially released. It has hurt a bit during the early months of 1.4 as we started dealing with various major issues. I would also like to note that the development team did recognize the difference in the situation we had at hand. These are the exact reasons we decided to fully maintain Asterisk 1.2 during the first 6 months of the life of 1.4. When Asterisk 1.2 was released, 1.0 was immediately deprecated and only maintained with security fixes. I am now feeling very good about Asterisk 1.4. When we had our developer conference in May, we talked about a lot of cool things. However, we also talked about how it must be a priority that we fix bugs and decided to work extremely hard on bugs for the Summer. We lived up to our word. The past few months have seen a *ton* of serious issues get resolved, and I am very pleased with our progress. To the whole user community, thank you very much for your support and patience with us as we push Asterisk forward. Feel confident that we will not leave you hanging. We will continue to do whatever we can to make Asterisk stable as we further improve functionality. If the changing needs of the user community mean that Asterisk 1.4 needs to be maintained for a full year after 1.6 is released, then so be it. Thanks for reading, -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error in loading libunicall.so module while running asterisk command
If you still have this problem, contact me via MSN at the same address I write from. Im sure that with 5 minutes in your box we can fix it. Regards On 8/24/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On 8/23/07, [EMAIL PROTECTED] Hi, I am using debian 4.0 with version 2.6.18-4-686 I have downloaded the required files form site asterisk-1.2.24.tar.gz libmfcr2-0.0.3-1.4.tar.bz2 libsupertone-0.0.2.tar.gz libunicall-0.0.3-1.4.tar.bz2 spandsp-20060903.tar.gz I downloaded and installed the files in the follwing sequence spandsp libsupertone libunicall libmfcr2-0.0.3 is giving a lot of definition error I converted .src.rpm file of libmfcr2 to .deb file and installed it. the copying the chn_unicall.c and channels_Makefile.patch to channels subdirectory of asterisk-1.2.24 but when I run ,asterisk -vvgc' on command line it gives following error message -- loader.c: 326 __load_resource:libunicall.so.0cannot open shared object filer loader.c:555load_modulesloading module chan_unicall.so failed but libunicall.so is present. Can you tell me how to trobleshoot it. Thanka and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
In article [EMAIL PROTECTED], Julian Lyndon-Smith [EMAIL PROTECTED] wrote: We have been running 1.4 since July 06 (it was trunk then), and have upgraded often with the 1.4 branch (Currently on SVN-branch-1.4-r77571). We have 100+ extensions (SIP) and 30 ISDN channels. We often have 50+ agents available for outbound calls and queues (20+ queues). We are making / receiving approx 5000+ calls per day. We use jabber and odbc heavily (updating / reading / Creating) as well as using odbc for cdr records. All calls are recorded (monitor at the moment). We use SMS inbound and outbound. This is on a dell 2850 with 2gb ram (top - 21:31:11 up 246 days). Asterisk has System uptime: 3 weeks, 4 days, 7 hours, 57 minutes, 44 seconds Whilst nowhere near the levels of some other people, for our purposes, 1.4 is working very very well for us, and the development guys have our gratitude and respect. It's a damn fine piece of work that has saved my company a lot of money in the 2 years we've been using asterisk. But it appears that you are not using IAX. I suspect until extremely recently it is IAX that has been the weak link in 1.4, because of the change from single-threaded to multi-threaded. The latest work on IAX with astobj2 looks like it should solve this at last! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk stable 1.2.x or 1.4.x
Dear all I am going to install asterisk on production now i m confused about version i dont know which version is good and best for my setup 1.2.x or 1.4.x can anyone tell me in detail which version whoud be best for my setup 1.4.x or 1.2.x if 1.4.x is good then which version whoud be better like 1.4.1 or 1.4.10 , 1.4.11 Regards satish patel - Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] which OS would be fine for asterisk
Dear which Linux version would be fine for asterisk CentOS 5.0 or Debian 4.0 or RHEL 4.0 Regards Satish patel - Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE120P digium card PRI_CPE error
Thnk now it is working fine according to your reply Regards satish patel Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Set it to pri_net instead of pri_cpe. IF you start getting error messages that We think we are NET and they think they are NET, then your carrier or the Avaya has the line in Loopback mode. satish patel wrote: Dear all I got one more error my asterisk E1 card connected with avaya E1 card [avaya]---E1-[asterisk] i got this 2 error what is start asteris on consol mode asterisk -c [Jul 27 09:51:29] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:30] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:31] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:32] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. 2-- second error [Jul 27 09:51:32] WARNING[737] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2007-021: Crash from invalid/corrupted MIME bodies when using voicemail with IMAP storage
Asterisk Project Security Advisory - AST-2007-021 ++ | Product | Asterisk | |+---| | Summary | Crash from invalid/corrupted MIME bodies when | || using voicemail with IMAP storage | |+---| | Nature of Advisory | Crash | |+---| | Susceptibility | Remote Unauthenticated Sessions | |+---| | Severity | minor | |+---| | Exploits Known | No| |+---| |Reported On | August 23, 2007 | |+---| |Reported By | Kevin Stewart | |+---| | Posted On | August 24, 2007 | |+---| | Last Updated On | August 24, 2007 | |+---| | Advisory Contact | Mark Michelson [EMAIL PROTECTED]| |+---| | CVE Name |CVE-2007-4521 | ++ ++ | Description | If Asterisk is configured to use IMAP as its backend | | | storage for voicemail, then an e-mail sent to a user | | | with an invalid/corrupted MIME body will cause Asterisk | | | to crash when the user listens to their voicemail using | | | the phone. | | | | | | This does not affect any other voicemail storage option, | | | nor does it affect users who check their voicemail via | | | e-mail when using IMAP storage. | ++ ++ | Resolution | Since this is a minor issue, a new release is not | || immediately planned. However, the issue will be fixed in | || Asterisk Open Source version 1.4.12 when it is released. | ++ ++ | Affected Versions| || |Product | Release | | || Series| | |+-+-| | Asterisk Open Source |1.0.x| Not Affected| |+-+-| | Asterisk Open Source |1.2.x| Not Affected| |+-+-| | Asterisk Open Source |1.4.x| Versions 1.4.5 - 1.4.11 | |+-+-| | Asterisk Business Edition|A.x.x| Not Affected| |+-+-| | Asterisk Business Edition|B.x.x| Not Affected| |+-+-| | AsteriskNOW | pre-release | Not Affected| |+-+-| | Asterisk Appliance Developer |0.x.x| Not Affected| | Kit | | | |+-+-| | s800i (Asterisk
Re: [asterisk-users] asterisk stable 1.2.x or 1.4.x
Hi On Fri, Aug 24, 2007 at 03:08:56PM -0700, satish patel wrote: Dear all I am going to install asterisk on production now i'm confused about version i dont know which version is good and best for my setup 1.2.x or 1.4.x Installing 1.2 on a new system now (that should hopefully be maintained for a while) may not be the best idea: you will need to spend more time looking for backported features and such. And new bugfixes will be harder to come by. This will be increasingly more true six monthes from now. Don't confuse this with a discussion regarding upgrade from 1.2 or not. Some people rightfully stated there if it's not broken don't fix it. However this is not the case for a new installation. can anyone tell me in detail which version whoud be best for my setup 1.4.x or 1.2.x if 1.4.x is good then which version whoud be better like 1.4.1 or 1.4.10 , 1.4.11 1.4.1? no way. Generally get the latest. Which would be 1.4.11. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Keeping queue counters after restarting
Hello, On 8/24/07, James FitzGibbon [EMAIL PROTECTED] wrote: Not sure about restarts, but trunk keeps them through reloads. How often are you restarting? My Asterisk has been segfaulting a few times during the day. I couldn't figure out why that's happening. safe_asterisk restarts Asterisk immediately, but all my calls are dropped and I lose the queue stats. I'll check that 'keepstats' option. Thanks. Regards. -- MARLON DUTRA Propus GnuPG ID: 0x3E2060AC pgp.mit.edu http://www.propus.com.br/ http://hackers.propus.com.br/~marlon/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Restart status
If I issue a restart gracefully command, the system will wait until all channels are idle before restarting. During the time the system is waiting for idle activity, is there a command that can let me know it is in graceful restart wait mode ? Thanks, Ron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 trunking scalability
Hi List, I have a 2Mbps SDSL link which gets saturated during peak time because about I have about 3 E1 worth of g729 traffic going thru. So I'm planning to use IAX2 trunking to reduce bandwith requirement and squeeze out each and every bit of this (expensive) bandwith. I've set up two boxes (debian etch), one in a remote data center (which has plenty of bandwith) and one behing the SDSL link. To make things consistent I've installed the same kernel, latest stable zapata + asterisk on both ends. I've done some tests with about 1/2 E1 (15 channels) worth of calls and so far it's been working good - and the call statistics (ASR, ACD, PDD) are roughtly the same. So far, so good! Now the big question is: how far can I expect it to scale? Has anybody successfully mounted IAX2 trunking with 3-4 E1s worth of traffic? Your experience and feedback is appreciated. Cheers, Jean-Michel. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as a softswitch
www.freeswitch.org http://www.freeswitch.org/ (still in early beta) _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Quitoriano Sent: Friday, August 24, 2007 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk as a softswitch What is a good softswitch that is also open source rather than asterisk? On 8/24/07, James Jones [EMAIL PROTECTED] wrote: Yes you could, but asterisk was designed to be a PBX. I would not use it as soft switch due its limitations. It really depends on how much traffic you are going to be passing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 24 August 2007 1:11 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk as a softswitch Mark Quitoriano wrote: Can i use asterisk as a softswitch? This thread has been discussed over and over. Search the archives, there are more thoughts and opinions there than you probably have time or desire to read. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.4/969 - Release Date: 23/08/2007 4:04 p.m. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.4/969 - Release Date: 23/08/2007 4:04 p.m. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/register http://www.spreadfirefox.com/?q=user/registerr=19441 r=19441 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help define the Asterisk regression test suite
On Fri, Aug 24, 2007 at 12:27:14PM -0500, Russell Bryant wrote: James FitzGibbon wrote: Let me ask a question myself: what kind of regression test does * undergo before release, and what level of traffic gets put through stuff like app_queue? I assume it's not real-world scale, else these hard to pin down concurrency issues we're seeing would have been caught in test. Let *me* ask a question. :) What level of heavy regression testing would you *expect* of an open source development team? No, let *me* ask a question: What tests *go* in a regression test suite for realtime switch software such as *? I've unthreaded this so we can all keep track of it; it should really be on a wiki somewhere, but I'm parochial: everything except Mediawiki gives me hives. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 trunking scalability
On 8/24/07, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Hi List, I have a 2Mbps SDSL link which gets saturated during peak time because about I have about 3 E1 worth of g729 traffic going thru. So I'm planning to use IAX2 trunking to reduce bandwith requirement and squeeze out each and every bit of this (expensive) bandwith. I've set up two boxes (debian etch), one in a remote data center (which has plenty of bandwith) and one behing the SDSL link. To make things consistent I've installed the same kernel, latest stable zapata + asterisk on both ends. I've done some tests with about 1/2 E1 (15 channels) worth of calls and so far it's been working good - and the call statistics (ASR, ACD, PDD) are roughtly the same. So far, so good! Now the big question is: how far can I expect it to scale? Has anybody successfully mounted IAX2 trunking with 3-4 E1s worth of traffic? Your experience and feedback is appreciated. So you are using an asterisk box as an E1 gateway. You want to know if switching from not using IAX trunking to using IAX trunking will have any effect? Yes it will lower your bandwidth usage a little. It will not increase the CPU load. If your system can support x calls it will be able to support the same amount of calls. The best thing you can do for your system is add a TC400B card. It will also legally support G723 codec which I think sounds just fine, but will save you a bit more bandwidth. Using the hardware transcoder will greatly increase the number of calls your system would be able to handle. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
On Fri, Aug 24, 2007 at 04:00:23PM -0400, Matt Florell wrote: With all of that said, I do have a testing setup that allows me to run tests at high loads on Asterisk, but not all scenarios can be checked in a testing setup. I ran a mid-volume test on 1.4.10 and it worked without crashing. I wanted to test a new feature in 1.4 so I put the server into production. It worked fine for a few hours under small load, but once the load increased there were several issues(mostly relating to stuck locks I am guessing) and the server would crash every few hours and also have some weird Manager API issues. So after a few days I rolled the server back to 1.2.X and all was well again. Running the tests again later at a higher call volume and on servers with more horsepower revealed the same crashes and other issues as I noticed in production. Here's a secondary question (and Matt, I *do* plan to get around the damned corner to one of your meetups one of these days :-): Just how easy is it to roll back to the older release when the feces hit the fan? Seems like making that simple would be pretty important? (Context: my boss is about to tip on playing with Asterisk, finally..) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which OS would be fine for asterisk
CentOS and RHEL are the same thing. One uses the RedHat trademark, the other doesnt. One is expensive, the other isn't. I don't like to recommend either because I just don't like RedHat's business practices. Personally I recommend SuSE Linux. OpenSuSE without the GUI installed will do just fine. If you want to buy SLES that's fine, but I really don't see the value in it. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gizmo revisited
Launched the OS X version of Gizmo after about a year of inactivity, downloaded the update and discovered the new improved Giszmo features Asterisk interoperability by allowing a secondary SIP account to be registered simultaneously. It also allows you to make the routing choice for outgoing calls; your own server or via Gizmo. So far, this is the best SIP softphone I've come across for OS X. It comes in other flavors and I thought I'd mention it as it can be free and I haven't seen it mentioned recently. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which OS would be fine for asterisk
I have used CentOS and it works fine and it is easy to install. I know that Debian is a little more complicated to install Asterisk and some teatures on Debian. I'd choice CentOS 4.2 or 4.4, as my personal preference. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
