[Asterisk-Users] FXO definition
Maybe I am to sensetive, but what is an FXO? I have a device in my hand, it says it has an FXS and FXO port (besides WAN and LAN port) The SIP settings are only effecting the FXS. The FXO is connected to the phone company but can only be reached from the phone connected to FXS by prepending a defined key (e.g. #) The FXO port is directly connected to FXS if the box is without power. A call from the phone company line will be directly connected to the phone on FXS. Can I still say it is an FXO port bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forbidden - wrong password on authentication for NOTIFY
How can I find out which phone and what is missing? WARNING[10532]: chan_sip.c:8669 handle_response: Forbidden - wrong password on authentication for NOTIFY bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need some statistics facts
There is nothing more than figures, right? I am looking for: 1. How many phone lines are currently in this world (I estimate 800 billion analog lines) 2. How many data lines are currently in this world (I estimate 1 billion) 3. what is the forcast of 1 2 ? When will it swap? 4. How many PBX are in use? I guess that a part will be extended with FXO to the Interent. Some will be replaced at all, because of missing features. 5. Which countries favour VoIP? Which countries forbid VoIP? Where are the cheapest / best ADSL lines available? Hongkong, Korea, Japanis my guess bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need some statistics facts
Dean Collins wrote: Ronald, Why? What do you need it for? For a power point slide. Would the statistic or the facts are different if I would need it for a report ? hehehehehe bye Ronald Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ronald_Wiplinger Sent: Tuesday, 9 August 2005 10:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Need some statistics facts There is nothing more than figures, right? I am looking for: 1. How many phone lines are currently in this world (I estimate 800 billion analog lines) 2. How many data lines are currently in this world (I estimate 1 billion) 3. what is the forcast of 1 2 ? When will it swap? 4. How many PBX are in use? I guess that a part will be extended with FXO to the Interent. Some will be replaced at all, because of missing features. 5. Which countries favour VoIP? Which countries forbid VoIP? Where are the cheapest / best ADSL lines available? Hongkong, Korea, Japanis my guess bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] realtime: sip show users/peers
I don't see anything with sip show users and sip show peers, however it works! Is there a trick? I have installed realtime (sipbuddies) on one machine and see sip show peers/users and on my new installed system I don't. Have I forgotten something? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] realtime: sip show users/peers
I don't see anything with sip show users and sip show peers, however it works! Is there a trick? I have installed realtime (sipbuddies) on one machine and see sip show peers/users and on my new installed system I don't. Have I forgotten something? Just seconds after I hit send, I found: if you enable RealTime caching in your sip.conf, Voicemail MWI works and so does 'sip show peers'. Can anybody explain me the other possible settings: ;rtnoupdate=yes ; do not send the update request over realtime. ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered when the registration expires ; the friend will vanish from the configuration until requested ; again. If set to an integer, friends expire ; within this number of seconds instead of the ; same as the registration interval ;rtignoreexpire=yes ; when reading a peer from Realtime, if the peer's registration ; has expired based on its registration interval, used the stored ; address information regardless bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some more VOICEMAILMAIN issue...
Mauro Zanin wrote:
Hi everybody,
I have corrected this line in extensions.conf by stripping spaces off
and now it executes:
*exten = 22999,1,VoiceMailMain(s${CALLERIDNUM})*
when it runs, the mail box number is asked and password too. I
expected no question were made, because I inserted CALLERIDNUMBER and
s in front of box number.
have you reloaded Asterisk?
bye
Ronald
Anybody knows why?
Thank to you all, very kind members of this list!
Ciao
Mauro
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[Asterisk-Users] Real-time for H.323?
Matthew, can we use real-time also for H.323 phones? (h323_buddies) ??? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 - two processes
Brian West wrote: If you use mp3nb from the sample configs you will have exactly 1 per class. Great! Where can I read more details about it? (musiconhold.conf) bye Ronald /b On Jul 26, 2005, at 9:38 PM, MF Hulber wrote: Yes, I always have two. MARK. Billy Dunn wrote: Does everyone have two processes running for mpg123? I always have them when I'm running an idle Asterisk box. No calls going in or out and nothing off hook. Is this normal? Thanks! 5008 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-ri 5015 ?S 0:00 /usr/sbin/asterisk 5061 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-ri ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can I use MySQL in the dialplan?
I would like to put / get some data from an MySQL database. I want to use this MySQL database also via a web page. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Keys ???
I would like to read more about Keys in asterisk. How to use it, how to create, bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP messengers video phones
Is there a possibility to send text based messages from/to a sip phone (text display) or to a video phone or from/to a messenger? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is soekris good?
asterisk_on_oelf wrote: Hi, I have a soekris 4801 since some days. I use it with a FritzCard-USB and an internal HFC-Card (NT Mode). Everything is working, but I still havn't had time for performance test. Only thing I tested, was two ISDN channels via FritzCard in a conference room. CPU usage was nearly 70% I hope next weekend I'll find more time. What WiFi phone do you want to use? I tried a ZyXEL P2000W, but voice quality was very bad. Jens, I am trying to find out what is the best board for us. I want to build an asterisk based PBX with one digium TDM422 card. A USB wireless adapter should make the entire system with: * [EMAIL PROTECTED] (Is @home or regular better?) * Shorwall firewall * QoS * Hotspot for wireless phones * web server for Asterisk (billing, settings) - maybe thttpd since it also can IPv6 * IPv6 in the second step (I think @home cannot IPv6) * astcc * h.323 module * wakeup * festival (Maybe the CPU / RAM is too low for that) * MOH * voice mail * ??? What do you think about it? Is the 4801 right for that? Some questions about Soekris: What is in the package? (Power adapter?, CF?, manual? ...) How to install it? What is the CF size you are using? and how much is still free? What have you installed? We are in the process to develop a WiFi phone, bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOWTO capture digits
J.Raborg wrote: Folks: does anybody have an idea? how to capture the DTMF digits to a file, after an extn asnwer? then POST it to a url? I guess you are looking for a sys command, e.g. echo bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Has anybody installed Sphinx?
Would like to exchange with Sphinx users, ... bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DBput from the web?
Ivan Meic (Vox Mundi) wrote: Does anybody has a php code for using DBput (DBget, DBdel) from a web interface, which database is used for astrisk? I don't have anything similar, but instead of using * internal DB maybe you should consider using MySQL. Do you have an example how to use MySQL in Asterisk? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to debug perl agi
Juan J. Sierralta P. wrote:
Hi,
syslog('info', 'hello Asterisk!');
That should go into the syslog for the facility user. It may end up on
/var/log/messages , /var/log/user.log or whereever your system sends
such log entries.
But why not print to STDERR? IIRC the stderr of AGI scripts goes to the
asterisk console.
Dunno but at least ASTCC as lot of prints to STDERR but none of
these appeared on my console.
As I can remember you get the print statements but NOT on a remote
console,
bye
Ronald
BTW I had to patch last ASTCC CVS since it wasn't getting the call time:
--- /home/juanjo/voip/astcc/astcc.agi 2005-07-11 03:28:06.0 -0400
+++ astcc.agi 2005-07-12 01:48:41.0 -0400
@@ -329,9 +329,10 @@
sub calccost() {
my ($adjconn, $adjcost, $answeredtime, $increment) = @_;
- eval { my $adjtime = int(($answeredtime + $increment - 1) /
$increment) * $increment };
+ my $adjtime = int(($answeredtime + $increment - 1) /
$increment) * $increment;
my $cost;
I'm using Perl 5.8:
[EMAIL PROTECTED]:~$ perl -v
This is perl, v5.8.4 built for i386-linux-thread-multi
Copyright 1987-2004, Larry Wall
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[Asterisk-Users] RTP not thru asterisk
I want to make sure that RTP is not going thru my asterisk. I read you should avoid in the dial commands: m music while ringing t,T transfer calls from caller and called party What else do I need to take care? remote phone === registered to local asterisk === calling remote gateway should have the RTP remote phone ===(RTP)== calling remote gateway bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to start with SER and Asterisk?
It is on my list for several months ;-( I want to have on my asterisk box also SER installed in the first step, whereby SER is nothing doing than just use Asterisk, if a feature of Asterisk is required. Calls from a phone to an ENUM phone number, or from sip to sip phone should not bother asterisk. Has anybody set this up, ... I understand that SER is only for SIP, is there a counterpart for IAX? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 still no success to dial out via GK
[public_gk]
;exten = _070.,1,Set(CALLERID(number)=07033${CALLERIDNUM})
exten = _070.,1,Dial(H323/[EMAIL PROTECTED])
exten = _070.,n,Hangup
*CLI h323 show peers
Name Accountcode ip:port Formats
7000 ast_h323 203.160.252.147:1720 0x4 (ulaw)
8867033 ast_h323 203.160.252.147:1720 0x4 (ulaw)
myfriend1ast_h323 a.b.c.d:1720 0x4 (ulaw)
mypeer1 ast_h323 a.b.c.d:1720 0x4 (ulaw)
*CLI
Jul 13 16:22:37 WARNING[15193]: chan_h323.c:914 h323_indicate: Don't
know how to indicate condition -1 on ooh323c_246
-- Executing Hangup(H323/Ronald Wiplinger (8770)-f01d, ) in new
stack
== Spawn extension (default, 070168170, 2) exited non-zero on
'H323/Ronald Wiplinger (8770)-f01d'
== No one is available to answer at this time (1:0/0/0)
.
My SIP phone gives me the ring tone, but the H323 site does not do anything.
Why? What do I do wrong?
bye
Ronald
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Re: [Asterisk-Users] Minutes Limits
Carlos Andres Fuentealba F. wrote: Hello everybody! someone knows how to limit minutes per month to a specific user? Use astcc. Than you can use one price for all calls and so you have money=time limit. If you want to make it silent replace all the used sound files with an empty one, bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Support needed
Will Velez wrote: Hi my name is Will Velez. Does Asterisk support E164? Thanks You can make your dialplan as you wish, I usually use the E.164 number, since it is easier to remember the number and easier to handle ENUM lookups than. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DBput from the web?
Does anybody has a php code for using DBput (DBget, DBdel) from a web interface, which database is used for astrisk? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Are chinese voice files available?
Are Chinese voice files available? (and if, where?) bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is soekris good?
We found at the wiki a link to soekris and wonder if it is good? Is anybody using it and can share some experience, please? We would like to use it as a small PBX including a wireless access point, so that we can also use WiFi phones. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asteriski misses the table
I am not aware what I have done wrong, but the result is a query of:
*Database error:* Invalid SQL: SELECT * FROM WHERE
UNIX_TIMESTAMP(calldate) = UNIX_TIMESTAMP('2005-07-01') ORDER BY
calldate DESC LIMIT 0,25
*MySQL Error*: 1064 (You have an error in your SQL syntax; check the
manual that corresponds to your MySQL server version for the right
syntax to use near 'WHERE UNIX_TIMESTAMP(calldate) =
UNIX_TIMESTAMP('2005-07-01') ORDER BY calldat' at line 1)
*Database error:* Invalid SQL: SELECT count(*) FROM WHERE
UNIX_TIMESTAMP(calldate) = UNIX_TIMESTAMP('2005-07-01')
*MySQL Error*: 1064 (You have an error in your SQL syntax; check the
manual that corresponds to your MySQL server version for the right
syntax to use near 'WHERE UNIX_TIMESTAMP(calldate) =
UNIX_TIMESTAMP('2005-07-01')' at line 1)
*Database error:* next_record called with no query pending.
*MySQL Error*: 1064 (You have an error in your SQL syntax; check the
manual that corresponds to your MySQL server version for the right
syntax to use near 'WHERE UNIX_TIMESTAMP(calldate) =
UNIX_TIMESTAMP('2005-07-01')' at line 1)
It misses the table.
I have set the table name in the /lib/defines.php
(I checked several times for a spelling error, quotemark, ...)
bye
Ronald
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[Asterisk-Users] Festival problems
I have installed festival a while ago and it could say Mary had a little lamb When I changed the text, it kept silence. Changed back the text and it worked. Now it does not say anything anymore!!! festival.conf: [general] usecache=yes cachedir=/var/cache/asterisk/festival/ festivalcommand=(tts_textasterisk %s 'file)(quit)\n ls -l /var/cache/asterisk/festival/ total 8 -rwxr-xr-x 1 root root 25 Jul 8 11:25 AD54B7C2BB46EBD196EFECC64330735D -rwxr-xr-x 1 root root 26 Jul 8 11:26 FA198D47557433B6B99E8AC6D3CCA6EB Note the file length of 8 *CLI -- Executing Answer(SIP/6002-a3f3, ) in new stack -- Executing Festival(SIP/6002-a3f3, mary had a little lamb) in new stack == Parsing '/etc/asterisk/festival.conf': Found What is wrong? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Video phone settings???
I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic canreinvite=yes context=from-sip callerid=Ronald Wiplinger 6003 dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Tests on 7/11/2005 Eybeam to 8770 both screens are black!!! e*CLI -- Executing Dial(SIP/6003-94ec, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6003-94ec -- SIP/6004-4b4d is ringing -- SIP/6004-4b4d answered SIP/6003-94ec -- Stopped music on hold on SIP/6003-94ec -- Attempting native bridge of SIP/6003-94ec and SIP/6004-4b4d == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6003-94ec' -- Eybeam to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6003-8a2e, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6003-8a2e -- SIP/6005-fa6a is ringing -- SIP/6005-fa6a answered SIP/6003-8a2e -- Stopped music on hold on SIP/6003-8a2e -- Attempting native bridge of SIP/6003-8a2e and SIP/6005-fa6a == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6003-8a2e' -- 8770 to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- 8770 to Eyebeam 8770 gets picture, Eybeam no picture -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- Executing Dial(SIP/6004-2cff, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6004-2cff -- SIP/6003-322c is ringing -- SIP/6003-322c answered SIP/6004-2cff -- Stopped music on hold on SIP/6004-2cff -- Attempting native bridge of SIP/6004-2cff and SIP/6003-322c == Spawn extension (from-sip, 6003, 1) exited non-zero on 'SIP/6004-2cff' -- 8882 to Eyebeam both screens are black!!! -- Executing Dial(SIP/6005-3361, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6005-3361 -- SIP/6003-9ed0 is ringing -- SIP/6003-9ed0 answered SIP/6005-3361 -- Stopped music on hold on SIP/6005-3361 -- Attempting native bridge of SIP/6005-3361 and SIP/6003-9ed0 -- 8882 to 8770 8882 gets a picture -- Executing Dial(SIP/6005-abd3, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6005-abd3 -- SIP/6004-6381 is ringing -- SIP/6004-6381 answered SIP/6005-abd3 -- Stopped music on hold on SIP/6005-abd3 -- Attempting native bridge of SIP/6005-abd3 and SIP/6004-6381 == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6005-abd3' Jul 11 15:34:27 WARNING[14974]: chan_sip.c:1046 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video phone settings???
Giorgio Incantalupo wrote: Hi, try videosupport=yes in the general section of sip.conf. Maybe it can work. I have already set that. Without that NO video at all at any try. bye Ronald Giorgio. Ronald_Wiplinger wrote: I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic canreinvite=yes context=from-sip callerid=Ronald Wiplinger 6003 dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Tests on 7/11/2005 Eybeam to 8770 both screens are black!!! e*CLI -- Executing Dial(SIP/6003-94ec, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6003-94ec -- SIP/6004-4b4d is ringing -- SIP/6004-4b4d answered SIP/6003-94ec -- Stopped music on hold on SIP/6003-94ec -- Attempting native bridge of SIP/6003-94ec and SIP/6004-4b4d == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6003-94ec' -- Eybeam to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6003-8a2e, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6003-8a2e -- SIP/6005-fa6a is ringing -- SIP/6005-fa6a answered SIP/6003-8a2e -- Stopped music on hold on SIP/6003-8a2e -- Attempting native bridge of SIP/6003-8a2e and SIP/6005-fa6a == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6003-8a2e' -- 8770 to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- 8770 to Eyebeam 8770 gets picture, Eybeam no picture -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- Executing Dial(SIP/6004-2cff, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6004-2cff -- SIP/6003-322c is ringing -- SIP/6003-322c answered SIP/6004-2cff -- Stopped music on hold on SIP/6004-2cff -- Attempting native bridge of SIP/6004-2cff and SIP/6003-322c == Spawn extension (from-sip, 6003, 1) exited non-zero on 'SIP/6004-2cff' -- 8882 to Eyebeam both screens are black!!! -- Executing Dial(SIP/6005-3361, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6005-3361 -- SIP/6003-9ed0 is ringing -- SIP/6003-9ed0 answered SIP/6005-3361 -- Stopped music on hold on SIP/6005-3361 -- Attempting native bridge of SIP/6005-3361 and SIP/6003-9ed0 -- 8882 to 8770 8882 gets a picture -- Executing Dial(SIP/6005-abd3, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6005-abd3 -- SIP/6004-6381 is ringing -- SIP/6004-6381 answered SIP/6005-abd3 -- Stopped music on hold on SIP/6005-abd3 -- Attempting native bridge of SIP/6005-abd3 and SIP/6004-6381 == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6005-abd3' Jul 11 15:34:27 WARNING[14974]: chan_sip.c:1046 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users
Re: [Asterisk-Users] Video phone settings???
Matt Riddell wrote: Ronald Wiplinger wrote: apenon apenon wrote: Yes I have faced with the same problem, try to upgrade your eyebeam, some old versions have problem. How to make the echo test? Just add a line to your extensions.conf: exten = 600,1,Echo() I tried this and it echos the picture back on Xten but not on the hard phones, ... bye Ronald And that should do it. Also try the hardphones with different resolutions/bandwidths (CIF/QCIF). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 and asterisk
We come into this section of the dialplan:
exten = 8867033,1,Wait(1)
exten = 8867033,n,SayUnixTime
exten = 8867033,n,NoOp(If you know the extension ...)
exten = 8867033,n,Dial(${PHONE_6003})
The caller from the GK hears only ringing, not the time.
The extension 6003 rings and I can pick up, but without any voice nor video.
athome*CLI
-- Executing Wait(H323/GVSC8770-e822, 1) in new stack
-- Executing SayUnixTime(H323/GVSC8770-e822, ) in new stack
-- Playing 'digits/day-2' (language 'en')
-- Playing 'digits/mon-6' (language 'en')
-- Playing 'digits/h-12' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/thousand' (language 'en')
-- Playing 'digits/5' (language 'en')
-- Playing 'digits/at' (language 'en')
-- Playing 'digits/11' (language 'en')
-- Playing 'digits/40' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/a-m' (language 'en')
-- Executing NoOp(H323/GVSC8770-e822, If you know the extension
...) in new stack
-- Executing Dial(H323/GVSC8770-e822, SIP/6003) in new stack
-- Called 6003
-- SIP/6003-cfe3 answered H323/GVSC8770-e822
-- Attempting native bridge of H323/GVSC8770-e822 and SIP/6003-cfe3
== Spawn extension (default, 8867033, 4) exited non-zero on
'H323/GVSC8770-e822'
What am I missing?
How can I call back to H323 via GK?
bye
Ronald
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[Asterisk-Users] h323 how to ?????
I try to get H323 to run, but have so far only partial success:
There is a Gatekeeper GK, where asterisk connects to.
The Gatekeeper sees Asterisk, and Asterisk sees the gatekeeper.
From the Network on the GK, asterisk is reachable via the number
07033. I have an extension on asterisk 6002, which is reachable.
I try to call a number attached to the gatekeeper (070168177) with the
dialing plan:
exten = _9070.,1,Set(CALLERID(number)=07033${CALLERIDNUM})
exten = _9070.,n,Dial(H323/${EXTEN:${TRUNKMSD}})
exten = _9070.,n,Hangup
CLI shows:
*CLI
-- Executing Set(SIP/6002-9fac, CALLERID(number)=070336002)
in new stack
-- Executing Dial(SIP/6002-9fac, H323/070168177) in new stack
-- Called 070168177
== No one is available to answer at this time (1:0/0/0)
-- Executing Hangup(SIP/6002-9fac, ) in new stack
== Spawn extension (from-sip, 9070168177, 3) exited non-zero on
'SIP/6002-9fac'
The gatekeeper sees nothing from that. I guess the syntax is wrong for
dialing. How should it be?
Video connection:
I try to call from an H323 soft phone through the gatekeeper to call the
extension 6003 (eyebeam)
H323 soft phone calls through GK Asterisk box without webcam installed:
-- Executing Dial(H323/203.160.252.147-a44c, SIP/8600) in new stack
Jul 8 13:51:37 WARNING[12674]: chan_sip.c:1742 create_addr: No such
host: 8600
Jul 8 13:51:37 NOTICE[12674]: app_dial.c:977 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Answer(H323/203.160.252.147-a44c, ) in new stack
-- Executing SetVar(H323/203.160.252.147-a44c, TIMEOUT(digit)=5)
in new stack
Jul 8 13:51:37 WARNING[12674]: pbx.c:5754 pbx_builtin_setvar_old:
SetVar is deprecated, please use Set instead.
-- Digit timeout set to 5
-- Executing SetVar(H323/203.160.252.147-a44c,
TIMEOUT(response)=10) in new stack
-- Response timeout set to 10
-- Executing BackGround(H323/203.160.252.147-a44c,
demo-congrats) in new stack
-- Playing 'demo-congrats' (language 'en')
== CDR updated on H323/203.160.252.147-a44c
-- Executing Dial(H323/203.160.252.147-a44c, SIP/6003|60|trm) in
new stack
-- Called 6003
-- Started music on hold, class 'default', on H323/203.160.252.147-a44c
-- SIP/6003-e756 is ringing
-- SIP/6003-e756 answered H323/203.160.252.147-a44c
-- Stopped music on hold on H323/203.160.252.147-a44c
-- Attempting native bridge of H323/203.160.252.147-a44c and
SIP/6003-e756
Jul 8 13:52:16 WARNING[12674]: chan_sip.c:3203 process_sdp: Unknown SDP
media type in offer: video 7156 RTP/AVP 105 34
Jul 8 13:52:16 WARNING[12674]: chan_h323.c:914 h323_indicate: Don't
know how to indicate condition 17 on ooh323c_1
Jul 8 13:52:21 WARNING[12674]: chan_h323.c:914 h323_indicate: Don't
know how to indicate condition 17 on ooh323c_1
No connection, not even audio!
sip.conf settings for 6003:
[6003]
type=friend
username=6003
secret=pwd
qualify=200
nat=yes
host=dynamic
canreinvite=yes
context=from-sip
callerid=Ronald Wiplinger 6003 ; Full caller
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=h261
allow=h263
allow=h263p
Xten's settings:
Enable this SIP account
Display name: Ronald at Leadtek
User name: 6003
Password: password
Authorization: 6003
Domain: 59.120.139.119
Domain Proxy:
x Register with domain
STUN server
x Manual override:stun.xten.com
Any hints are welcome
bye
Ronald
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Re: [Asterisk-Users] No sound
Julio Cesar Ody wrote: Thanks all of you. I forgot to set the stun server, bye Ronald So you mean your SIP server is on the outside network and your peers on the private one? If so, sound behind NAT is always an issue, because it goes through a different port than 5060 udp (which is the reg. port). I had the same problem a few days ago. Posted the question to the list twice, and found help in the #asterisk channel at irc.freenode.net. Probably they can help you as well. But give www.voip-info.org a shot. Search fot NAT related issues. On 7/7/05, Ronald Wiplinger [EMAIL PROTECTED] wrote: I have an asterisk box installed, but all connections to outside of the private network do not have a sound. Can you give me a hint what it could it be? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
Robert Goodyear wrote: Well if you say it's registered, then packets are getting to asterisk and asterisk is accepting them, and you've allowed that SIP client. So... if you say there's absolutely NOTHING happening when the phone dials, then it sure seems like the phone is bad -- again, assuming no event whatsoever is happening when you dial. What else have you done to debug this? Have you registered the phone directly against another * box? Have you registered another phone against this * box? *CLI sip show peers Name/username HostDyn Nat ACL Mask Port Status 6002/6002 10.10.10.126 D N 255.255.255.255 1720 OK (58 ms) 6001/6001 10.10.10.125 D N 255.255.255.255 5061 OK (5 ms) 2 sip peers [2 online , 0 offline] if I dial from one phone to the other, sometimes it goes throu, sometimes the phone just keep silent and NOTHING is shown in the *CLI prompt Since it is from both phones, which work on another asterisk box, I can be sure, that the phones are working. The network is just two phones, one hub and to the asterisk box. home*CLI show version Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-06-28 04:53:43 bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ENUM
I try to use ENUM lookup ...
[trunkint]
exten = _9011Z.,1,SetCIDNum(${CALLERIDNUM}|a)
exten = _9011Z.,n,EnumLookup(${EXTEN:4})
exten = _9011Z.,n,BackGround(enum-lookup-successful)
exten = _9011Z.,n,Dial(${ENUM},30)
exten = _9011Z.,n,Hangup
*CLI
-- Executing SetCIDNum(SIP/6002-8607, 6002|a) in new stack
-- Executing EnumLookup(SIP/6002-8607, 886228357765) in new stack
-- Executing BackGround(SIP/6002-8607, enum-lookup-successful)
in new stack
-- Playing 'enum-lookup-successful' (language 'en')
-- Executing Dial(SIP/6002-8607, |30) in new stack
Jul 5 15:50:24 WARNING[22297]: app_dial.c:694 dial_exec_full: Dial
argument takes format
(technology1/number1technology2/number2...|optional timeout)
== Spawn extension (from-sip, 9011886228357765, 4) exited non-zero on
'SIP/6002-8607'
What have I forgotten?
bye
Ronald
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[Asterisk-Users] ENUM
I got one step further by editing enum.conf and use also
search = e164.org
Now the called party pickus up, but no voice can be heard at neither sides.
sip.conf:
[6002]
type=friend
username=6002
secret=secret
qualify=200
nat=yes
host=dynamic
canreinvite=yes
context=from-sip
callerid=Ronald Wiplinger 6002
dtmfmode=rfc2833
disallow=all
ulaw
allow=ulaw
allow=alaw
I try to use ENUM lookup ...
[trunkint]
exten = _9011Z.,1,SetCIDNum(${CALLERIDNUM}|a)
exten = _9011Z.,n,EnumLookup(${EXTEN:4})
exten = _9011Z.,n,BackGround(enum-lookup-successful)
exten = _9011Z.,n,Dial(${ENUM},30)
exten = _9011Z.,n,Hangup
*CLI
-- Executing SetCIDNum(SIP/6002-8607, 6002|a) in new stack
-- Executing EnumLookup(SIP/6002-8607, 886228357765) in new stack
-- Executing BackGround(SIP/6002-8607, enum-lookup-successful)
in new stack
-- Playing 'enum-lookup-successful' (language 'en')
-- Executing Dial(SIP/6002-8607, |30) in new stack
Jul 5 15:50:24 WARNING[22297]: app_dial.c:694 dial_exec_full: Dial
argument takes format
(technology1/number1technology2/number2...|optional timeout)
== Spawn extension (from-sip, 9011886228357765, 4) exited non-zero on
'SIP/6002-8607'
What have I forgotten?
bye
Ronald
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Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
Robert Goodyear wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ...yet no other information in the CLI or logs? C'mon, help us help you. The clue is in the question. I cannot make up a CLI entry ;-) There is nothing about it!!! As I said it is like it is not connected! bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: asterisk strips off trailing digit from incoming calls
Bernie Ott wrote:
There's a tiny bit of new info available:
asterisk only strips off the trailing digit of calls coming from
ANALOG lines; calling from e.g. a (fully digital) gsm mobile, I get
the 3 digit extension as it should be.
Try an extension with four digits and one with two. You may see, that *
chops all after 10 digits!!!
bye
Ronald
does this ring a bell for anyone?
On 7/3/05, no name [EMAIL PROTECTED] wrote:
so here it is, the problem that's been nagging me for the past 2 days:
connected a box to my telco's NTBA - zap/asterisk. which works:
box:/etc/asterisk# cat /proc/zaptel/1
Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7) HDB3/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
so then I instructed asterisk to treat this as zap interface:
box:/etc/asterisk# egrep -v '(^;|^$)' zapata.conf
[channels]
switchtype=euroisdn
signalling=bri_cpe
pridialplan=unknown
prilocaldialplan=unknown
immediate=no
priindication=outofband
overlapdial=no
usecallerid=yes
rxgain=0.0
txgain=0.0
context=inbound
callerid=asreceived
group=1
channel=1-2
defined this in the dialplan:
office:/etc/asterisk# tail -21 extensions.conf| egrep -v '(^;|^$)'
[inbound]
; my main number is 1234567,
; I am using 3-digit internal extensions
exten = _11234567XXX,1,Goto(internal-phones,$EXTEN:8,1)
exten = _XXX,1,Goto(internal-phones,${EXTEN},1)
; this acts as catch-all so dialling just the main number goes to x200
exten = s,1,Answer
exten = s,2,Goto(internal-phones,200,1)
now when I call e.g. 1234567200 from the outside, asterisk sees this as:
-- Extension '20' in context 'inbound' from '1some other number'
does not exist. Rejecting call on channel 0/1, span 1
why does asterisk INSIST on chopping the trailing digit off the
dialled number? I don't get it.
please help!
Bernie
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Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
Robert Goodyear wrote: On Jul 2, 2005, at 8:33 PM, Ronald Wiplinger wrote: Robert Goodyear wrote: On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sometimes yes - sometimes no (dialplan)
I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. What could be the reason? I have installed Festiva, and was only able once to listen a text to speech, since then this extension number never gives me a tone. Sometimes it shows up in the CLI, but without a tone on the phone. Other extensions have the same... bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GnuGK and Asterisk
I want to use Asterisk registering itself to a GK. SIP phones are registered to Asterisk H323 are registered to the GK I want to: 1. make calls from SIP (Asterisk) -- H323 (GK) 2. use Meetme to make a conference call for both types of phones I got on the GK, login and password, IP of GK, and codex g711u How to set-up h323.conf and extensions.conf for that? (I am using ooh323c) bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323
Tzafrir Cohen wrote: On Tue, Jun 28, 2005 at 01:18:18PM +0800, Ronald_Wiplinger wrote: Can anybody give me a hint, what I am doing wrong, please? Asterisk and H.323 1. download all parts - wget http://www.inaccessnetworks.com/asterisk-oh323/download/asterisk-oh323-0.7.1.tar.gz What version of Asterisk? I suppose you use HEAD, right? 0.7 requires HEAD. It is head! wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323_1.12.2.tar.gz wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib_1.5.2.tar.gz Isn't that an old version of pwlib and oh323? In the README file is another version, which I cannot find, however on the web they say: this are the files we compiled with wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/ohphone_1.4.1.tar.gz wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz WHIW, the debian debs in Sarge are of Asterisk 1.0, oh323 0.66pre3, openh323 1.1.15 and pwlib 1.8.4 . They build. IIRC I even got some reports that they work. what is the channel for that right? It seems that with each try (remove and make it new) I am not anymore coming that far as with the try before ;-( bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: H323
Tony Mountifield wrote: I corrected according to your suggestion: wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz These are not just patches, they are the complete source at patchlevel 4. You use these instead of the old versions mentioned previously. and in asterisk-oh323-0.7.1 I get: # make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/wrapper' make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory chan_oh323.c: In function `oh323_exception': chan_oh323.c:1145: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_indicate': chan_oh323.c:1326: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_digit': chan_oh323.c:1388: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_text': chan_oh323.c:1410: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_call': chan_oh323.c:1434: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1602: error: dereferencing pointer to incomplete type chan_oh323.c:1710: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_read': chan_oh323.c:1738: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_write': chan_oh323.c:2039: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_answer': chan_oh323.c:2231: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_fixup': chan_oh323.c:2275: error: dereferencing pointer to incomplete type chan_oh323.c: In function `ast_oh323_new': chan_oh323.c:2507: error: dereferencing pointer to incomplete type chan_oh323.c:2516: error: dereferencing pointer to incomplete type chan_oh323.c:2518: error: dereferencing pointer to incomplete type chan_oh323.c:2525: error: dereferencing pointer to incomplete type chan_oh323.c:2526: error: dereferencing pointer to incomplete type chan_oh323.c:2527: error: dereferencing pointer to incomplete type chan_oh323.c:2528: error: dereferencing pointer to incomplete type chan_oh323.c:2529: error: dereferencing pointer to incomplete type chan_oh323.c:2530: error: dereferencing pointer to incomplete type chan_oh323.c:2531: error: dereferencing pointer to incomplete type chan_oh323.c:2532: error: dereferencing pointer to incomplete type chan_oh323.c:2533: error: dereferencing pointer to incomplete type chan_oh323.c:2534: error: dereferencing pointer to incomplete type chan_oh323.c:2535: error: dereferencing pointer to incomplete type chan_oh323.c:2536: error: dereferencing pointer to incomplete type chan_oh323.c:2537: error: dereferencing pointer to incomplete type chan_oh323.c:2538: error: dereferencing pointer to incomplete type chan_oh323.c:2539: error: dereferencing pointer to incomplete type chan_oh323.c:2540: error: dereferencing pointer to incomplete type chan_oh323.c:2541: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_request': chan_oh323.c:2713: error: dereferencing pointer to incomplete type chan_oh323.c:2715: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_atexit': chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type chan_oh323.c: In function `load_module': chan_oh323.c:5192: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_oh323.c:5192: error: too many arguments to function `ast_channel_register' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/asterisk-driver' make: *** [subdirs_build] Error 1 find / -name channel_pvt.h -printdoes not return anything ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: H323
Ronald_Wiplinger wrote: Tony Mountifield wrote: I corrected according to your suggestion: wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz These are not just patches, they are the complete source at patchlevel 4. You use these instead of the old versions mentioned previously. and in asterisk-oh323-0.7.1 I get: # make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/wrapper' make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory chan_oh323.c: In function `oh323_exception': chan_oh323.c:1145: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_indicate': chan_oh323.c:1326: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_digit': chan_oh323.c:1388: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_text': chan_oh323.c:1410: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_call': chan_oh323.c:1434: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1602: error: dereferencing pointer to incomplete type chan_oh323.c:1710: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_read': chan_oh323.c:1738: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_write': chan_oh323.c:2039: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_answer': chan_oh323.c:2231: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_fixup': chan_oh323.c:2275: error: dereferencing pointer to incomplete type chan_oh323.c: In function `ast_oh323_new': chan_oh323.c:2507: error: dereferencing pointer to incomplete type chan_oh323.c:2516: error: dereferencing pointer to incomplete type chan_oh323.c:2518: error: dereferencing pointer to incomplete type chan_oh323.c:2525: error: dereferencing pointer to incomplete type chan_oh323.c:2526: error: dereferencing pointer to incomplete type chan_oh323.c:2527: error: dereferencing pointer to incomplete type chan_oh323.c:2528: error: dereferencing pointer to incomplete type chan_oh323.c:2529: error: dereferencing pointer to incomplete type chan_oh323.c:2530: error: dereferencing pointer to incomplete type chan_oh323.c:2531: error: dereferencing pointer to incomplete type chan_oh323.c:2532: error: dereferencing pointer to incomplete type chan_oh323.c:2533: error: dereferencing pointer to incomplete type chan_oh323.c:2534: error: dereferencing pointer to incomplete type chan_oh323.c:2535: error: dereferencing pointer to incomplete type chan_oh323.c:2536: error: dereferencing pointer to incomplete type chan_oh323.c:2537: error: dereferencing pointer to incomplete type chan_oh323.c:2538: error: dereferencing pointer to incomplete type chan_oh323.c:2539: error: dereferencing pointer to incomplete type chan_oh323.c:2540: error: dereferencing pointer to incomplete type chan_oh323.c:2541: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_request': chan_oh323.c:2713: error: dereferencing pointer to incomplete type chan_oh323.c:2715: error: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_atexit': chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type chan_oh323.c: In function `load_module': chan_oh323.c:5192: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_oh323.c:5192: error: too many arguments to function `ast_channel_register' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/asterisk-driver' make: *** [subdirs_build] Error 1 find / -name channel_pvt.h -printdoes not return anything I found a hint to steal it from the current version, but even that gives me an error: make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o make[1]: Leaving directory `/usr/src
[Asterisk-Users] H323
Can anybody give me a hint, what I am doing wrong, please? Asterisk and H.323 1. download all parts - wget http://www.inaccessnetworks.com/asterisk-oh323/download/asterisk-oh323-0.7.1.tar.gz wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323_1.12.2.tar.gz wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib_1.5.2.tar.gz wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/ohphone_1.4.1.tar.gz wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz wget http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz and untar it into /usr/src/ 2. Compiling cd /usr/src/pwlib pwlib$ ./configure pwlib$ make clean; make opt cd /usr/src/openh323 openh323$ patch -p1 /usr/src/asterisk-oh323-0.7.1/openh323_1.13.5-make.patch openh323$ ./configure openh323$ make clean; make opt 3. Download / update Asterisk - cd /usr/src export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login- the password is anoncvs. cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds go into each directory and: make clean; make update; make install Then, edit Makefile inside the asterisk-oh323-x.x.x directory and set the paths/options according to your system: DESTDIR=/ PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 ASTERISKINCDIR=/usr/src/asterisk/include ASTERISKMODDIR=/usr/lib/asterisk/modules ASTERISKETCDIR=/etc/asterisk OH323WRAPLIBDIR=/usr/local/lib SSLINCDIR=/usr/include/openssl SSLLIBDIR=/usr/lib Type make to build the oh323wrap library and the ASTERISK OH323 channel driver. [EMAIL PROTECTED] asterisk-oh323-0.7.1]# make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done /usr/src/openh323/openh323u.mak:296: warning: overriding commands for target `ccflags' /usr/src/openh323/openh323u.mak:293: warning: ignoring old commands for target `ccflags' /usr/src/openh323/openh323u.mak:296: warning: overriding commands for target `ccflags' /usr/src/openh323/openh323u.mak:293: warning: ignoring old commands for target `ccflags' make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 g++ -DP_LINUX=2.6.5-1.358smp -ffunction-sections -fdata-sections -D_REENTRANT -Wall -fPIC -DP_USE_PRAGMA -DPHAS_TEMPLATES -I/usr/src/pwlib/include/ptlib/unix -I/usr/include/pwlib -I/usr/src/pwlib/include -DPTRACING -I/usr/src/openh323/include -DHAS_IXJ -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.5.2\ -DOPENH323VERSION=\1.12.2\ -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -I/usr/src/openh323/include/openh323 -I../asterisk-driver -c asteriskaudio.cxx -o asteriskaudio.o asteriskaudio.cxx: In destructor `virtual PAsteriskSoundChannel::~PAsteriskSoundChannel()': asteriskaudio.cxx:167: error: `baseChannel' undeclared (first use this function) asteriskaudio.cxx:167: error: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: *** [asteriskaudio.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/wrapper' make: *** [subdirs_build] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
