Quantcast
Channel: Asterisk Forums
Viewing all 7460 articles
Browse latest View live
↧

asterisk 13 confbridge recording

November 14, 2014, 2:44 pm
$
0
0
sent to the user list as well:

We upgraded from asterisk 11 to asterisk 13. Recordings were working fine in 11 but nothing is being written on 13.



Here is the dialplan segment



same => n,ExecIF($["${TL_PHONE_CALL_RECORD}"="TRUE"]?SET(CONFBRIDGE(bridge,record_conference)=yes))

same => n,ExecIF($["${TL_PHONE_CALL_RECORD}"="TRUE"]?SET(CONFBRIDGE(bridge,record_file)=/var/spool/asterisk/confbridge/${TL_PHONE_CALL_ID}.wav))

same => n,ConfBridge(${TL_PHONE_CALL_ID},default_bridge,TTM_caller,TTM_caller_menu)



Here is the log from asterisk 13



-- Executing [s@TTMConferenceTalker:6] ExecIf("SIP/sbc1-00002426", "1?SET(CONFBRIDGE(bridge,record_conference)=yes)") in new stack

-- Executing [s@TTMConferenceTalker:7] ExecIf("SIP/sbc1-00002426", "1?SET(CONFBRIDGE(bridge,record_file)=/var/spool/asterisk/confbridge/296955.wav)") in new stack

-- Executing [s@TTMConferenceTalker:8] ConfBridge("SIP/sbc1-00002426", "296955,default_bridge,TTM_profile,TTM_profile_menu") in new stack

-- Channel SIP/sbc1-00002426 joined 'softmix' base-bridge <f14cbf31-d82e-4e2f-9c62-3f70b2c66165>



Here is a sample log from asterisk 11



[Oct 27 15:08:48] VERBOSE[14718][C-0005278d] pbx.c: -- Executing [s@TTMConferenceCaller:5] ExecIf("SIP/sbc1-0004e882", "1?SET(CONFBRIDGE(bridge,record_conference)=yes)") in new stack

[Oct 27 15:08:48] VERBOSE[14698][C-0005278c] pbx.c: -- Executing [s@TTMConferenceTalker:6] ExecIf("SIP/sbc1-0004e881", "1?SET(CONFBRIDGE(bridge,record_conference)=yes)") in new stack

[Oct 27 15:08:48] VERBOSE[14718][C-0005278d] pbx.c: -- Executing [s@TTMConferenceCaller:6] ExecIf("SIP/sbc1-0004e882", "1?SET(CONFBRIDGE(bridge,record_file)=/var/spool/asterisk/confbridge/278731.wav)") in new stack

[Oct 27 15:08:48] VERBOSE[14718][C-0005278d] pbx.c: -- Executing [s@TTMConferenceCaller:7] ConfBridge("SIP/sbc1-0004e882", "278731,default_bridge,TTM_caller,TTM_caller_menu") in new stack

[Oct 27 15:08:48] VERBOSE[14698][C-0005278c] pbx.c: -- Executing [s@TTMConferenceTalker:7] ExecIf("SIP/sbc1-0004e881", "1?SET(CONFBRIDGE(bridge,record_file)=/var/spool/asterisk/confbridge/278731.wav)") in new stack

[Oct 27 15:08:48] VERBOSE[14698][C-0005278c] pbx.c: -- Executing [s@TTMConferenceTalker:8] ConfBridge("SIP/sbc1-0004e881", "278731,default_bridge,TTM_profile,TTM_profile_menu") in new stack

[Oct 27 15:08:48] VERBOSE[14730] app_mixmonitor.c: == Begin MixMonitor Recording ConfBridgeRecorder/conf-278731-uid-1917613225

Statistics : Posted by lahoma • on Fri Nov 14, 2014 4:44 pm • Replies 0 • Views 4
↧
Search

What's wrong in my asterisk configuration ?

November 13, 2014, 1:55 pm
$
0
0
_. is unsafe because it also matches the special extension like h.

Statistics : Posted by gregchelli • on Thu Nov 13, 2014 3:55 pm • Replies 17 • Views 368
↧
↧

1200 SIP Extensions What you suggest.

November 15, 2014, 10:33 am
$
0
0
Hi Friends,
I am having 1200 SIP extensions. What you suggest should I keep all these on a single server or do you suggest some load balancing?
Please share your experience if you had such large number of extensions.

Statistics : Posted by numan82 • on Sat Nov 15, 2014 12:33 pm • Replies 0 • Views 26
↧

Setting up Small project

November 15, 2014, 12:28 pm
$
0
0
Hello people,

Im really new to all this and Ive been trying to set up a small VOIP with asterisknow. Ive set up asterisknow on a virtuarmachine VMware, also I have a switch which is a enterasys and a few Mitel IP Phones.

So far Ive been able to make a softphone call another with the static IP I set on asterisknow (ip address 192.168.1.140 subnetmask 255.255.255.0 Default Gateway 192.168.1.1). I created two extentions on the freepbx GUI and it worked fine on the same computer. Now that I want to connect to a phisical phone I cant find how to do this. Ive configured the phone to have the ip 192.168.1.141 same mask same default gw.

As for the enterasys switch I gave it the next config
ip address 192.168.1.145
Also I created a VLan and included all the fe from 1 to 10 (it has 48)
As extra data I have the computer were asterisknow is running with the ip address 192.168.1.139

I got the phone on port 2 the computer(with vmware and asterisknow) on port 3 and a computer (where im setting up the switch) on port 2

Im lost from here, I dont know what Im suppose to do.

I tried pinging from the computer with the switch configuration to the phone and it works. But when I try pingin asterisknow it wont respong but also when I try pinging the computer were asterisknow is the ping is recived. I dont know why this is happening. Could anyone tell me a good tutorial or something I can look at to connect the phisical phone to asterisk now and configuring correctly the switch or my computer so I can ping asterisknow too.

if you have any questions please do ask Image

Thanks in advance to anyone who helps Image

Statistics : Posted by jfeuchter • on Sat Nov 15, 2014 2:28 pm • Replies 0 • Views 27
↧

Asterisk and Sipgate.co.uk Outbound Calls Not Working

November 15, 2014, 12:49 pm
$
0
0
Hi
I am trying to create a Lync 2013 lab setup at home. For the purpose of attempting to create real world scenarios I wanted my lync lab to break out to the PSTN. Normally on production Lync deployments I have another guy with me who provides the config for the voice gateway systems which are near always Cisco or Avaya. We have never done an Asterisk Gateway before so this is new to me.

I have created an account with sipgate to get a free SIP trunk and I have successfully connected * to the sipgate service as I can see it online in the sipgate control panel.

I am able to make calls from the PSTN to Lync extensions without any trouble

However, I cannot call the PSTN from Lync Extensions. I can see the SIP Invite hitting the * box so I know Lync is working fine and the issue is with *.

I see and hear the following errors

Hear = All Circuits Are Busy Now (this comes from * box)
See = tcptls.c:863 ast_tcptls_client_start: Unable to connect SIP socket to 217.10.79.23:5060: Connection timed out in the logs

This confuses me because I have configured the trunks to use tcp and not tls.

In my outbound routes I have a dial plan for X. to route via the SIPGATE Trunk and another outbound Route which matches my SIPGATE user ID to route to Lync

The context from the lync trunk is from-internal and the context for the Sipgate trunk is from-trunk

Weirdly I tried * because when I tried Elastix I could dial out to the PSTN from Lync but could not receive calls from the PSTN.

Can anyone shed any light on where I have gone wrong?
thanks

Statistics : Posted by mve83 • on Sat Nov 15, 2014 2:49 pm • Replies 0 • Views 20
↧
↧

asterisk 11.5 with postgres realtime

November 15, 2014, 1:49 pm
$
0
0
Hello,
I tried to install and add users and session info via asterisk realtime module on postgresql

First i tried to create the example db schemas from various webpages under a database named "hsta"

Anyway, first i created the cdr_pgsql.conf file
with the following information
[global]
hostname=127.0.0.1
port=5598
dbname=hsta
password=secret
user=postgres
table=cdr

I confirmed that it is logging on db server fine..
Second step was to make sip_peers work on sql service.
extconfig.conf

[settings]
sipusers => pgsql,general,sip_conf
sippeers => pgsql,general,sip_conf

Then res_pgsql.conf
[general]
dbhost=127.0.0.1
dbport=5598
dbname=hsta
dbuser=postgres
dbpass=secret
requirements=warn


On sip.conf i have only general context.. I couldn find out how to add a realtime switch to this part.
Any help will be great..
When i check my sql, i see no select to load users from db, nor i cant see any users on sip show peers too.

ps: also sip.conf
[general]
context=from-sip

and extensions.conf:
[from-sip]
switch =>Realtime

Statistics : Posted by gobris • on Sat Nov 15, 2014 3:49 pm • Replies 0 • Views 20
↧

Setting up Small project

November 15, 2014, 12:28 pm
$
0
0
Is there any chance of doing this without FreePBX? This is the wrong place for FreePBX support and the FreePBX dialplan can make debugging difficult.

Please provide logs at at least verbosity 5. You may need to provide SIP debug logs, later. If you are able to eliminate FreePBX, please provide extensions.conf, and sip.conf. If not, please provide sip.conf and the transitive closure of what it includes (everything that can be found by following #includes, at every level, starting from there).

Statistics : Posted by jfeuchter • on Sat Nov 15, 2014 2:28 pm • Replies 1 • Views 82
↧

Transfer call to a Queue in PHPAGI

November 16, 2014, 6:42 am
$
0
0
Hi,

I am having difficulty transferring a call to a queue through PHPAGI. I am using the
Code: $agi->exec('Queue',"1000");
but the call keeps looping and the static agents never ring. Eventually I have to then restart the Asterisk.

Any help would be appreciated.

I know we can do it through the dialplan, but the situation requires me to do it from the AGI script.

Thanks in advance.
Jeet.

Statistics : Posted by jeet • on Sun Nov 16, 2014 8:42 am • Replies 0 • Views 43
↧
Search

Asterisk and Sipgate.co.uk Outbound Calls Not Working

November 15, 2014, 12:49 pm
$
0
0
Please provide your configuration.

Note TLS is TCP, but it is encrypted TCP and has a different default port. I can't think of any reason you would want to use TCP with sipgate if it weren't encrypted.

Statistics : Posted by mve83 • on Sat Nov 15, 2014 2:49 pm • Replies 5 • Views 130
↧
↧

Asterisk and Sipgate.co.uk Outbound Calls Not Working

November 15, 2014, 12:49 pm
$
0
0
How do I get asterisk to connect on TCP then because in the sip settings I have tcpenable as yes?

Statistics : Posted by mve83 • on Sat Nov 15, 2014 2:49 pm • Replies 5 • Views 130
↧

Asterisk and Sipgate.co.uk Outbound Calls Not Working

November 15, 2014, 12:49 pm
$
0
0
It would be unusual to support encrypted connections on port 5060, which is what you are trying to do.

Statistics : Posted by mve83 • on Sat Nov 15, 2014 2:49 pm • Replies 5 • Views 130
↧

Asterisk and Sipgate.co.uk Outbound Calls Not Working

November 15, 2014, 12:49 pm
$
0
0
Hi

I think they do because I can call out of elastix but not asterisk

Statistics : Posted by mve83 • on Sat Nov 15, 2014 2:49 pm • Replies 5 • Views 130
↧

Asterisk and Sipgate.co.uk Outbound Calls Not Working

November 15, 2014, 12:49 pm
$
0
0
Maybe sipgate don't support encrypted connections, or maybe they don't support any kind of TCP connection.

Statistics : Posted by mve83 • on Sat Nov 15, 2014 2:49 pm • Replies 5 • Views 130
↧
↧

Cannot blind transfer call from SIP to SIP asterisk 12.7.0

November 16, 2014, 3:31 pm
$
0
0
Hi

I would like to be able to blind transfer from one local sip extension to another but this fails.

I have the following setup.

Dialing out to the PSTN with a SIP server, with a Dial command that has T (calling party enabled transfers enabled).

In test 1, I use an IAX2 clients connected to the asterisk 12.7.0 machine. When I press ## (which is what I have configured blind transfers) I get a ringtone, and then I enter an extension, and it works. ie. the call is dropped from the originating extension and the other extension rings.

When I do the above with two sip clients locally, I press ## then I get the rington, I enter an extension and then it returns me to the call. I checked in the source code bridge.c file and found that it was finding the extension correct but it wasn't doing the blind transfer.

So I changed line 4276 (or thereabouts) to force it do to a transfer if (1 ||do_bridge_transfer) and everything works fine.

Is there some configuration option that I am missing so that this flag isn't set to 1 in
4262 transfer_prohibited = ast_test_flag(&bridge->feature_flags,
4263 AST_BRIDGE_FLAG_TRANSFER_PROHIBITED);
4264 do_bridge_transfer = ast_test_flag(&bridge->feature_flags,
4265 AST_BRIDGE_FLAG_TRANSFER_BRIDGE_ONLY) ||
4266 ao2_container_count(channels) > 2;

Any help would be appreciated,

Cheers

Chiwong

Statistics : Posted by chi888 • on Sun Nov 16, 2014 5:31 pm • Replies 0 • Views 14
↧

Trouble getting asterisk running on Server2012 R2 64 bit

November 17, 2014, 8:38 am
$
0
0
I wasn't aware that any version was supported on Windows. The original Windows port is based on a version that hit end of life many years ago.

Statistics : Posted by aeldridge • on Mon Nov 17, 2014 10:38 am • Replies 3 • Views 78
↧
Search

Trouble getting asterisk running on Server2012 R2 64 bit

November 17, 2014, 8:38 am
$
0
0
I am getting the error below evertime i try and run Asterisk. Thanks for help in advance.


$ /cygdrive/d/infoscan/server/asterisk/modules/asterisk

Loading Asterisk module...
Asterisk module loaded successfully
Asterisk entry point foundMaster config file used: '/cygdrive/d/InfoScan/Server/
asterisk/etc/asterisk.conf'
Nov 17 11:16:52 NOTICE[1996]: cdr.c:1229 do_reload: CDR simple logging enabled.
Registered tone zone 0 (United States / North America)
Nov 17 11:16:53 ERROR[1996]: win32_tapi.c:129 telephonyInitialize: No suitable l
ine found.
Nov 17 11:16:54 NOTICE[1996]: win32_tapi.c:639 TapiEventThread: Terminating TAPI
msg thread...
Nov 17 11:16:54 ERROR[1996]: chan_tapi.c:870 load_module: Unable initialize TAPI

Nov 17 11:16:54 WARNING[1996]: loader.c:416 __load_resource: chan_tapi.so: load_
module failed, returning -1
Nov 17 11:16:54 NOTICE[1996]: win32_tapi.c:237 telephonyShutdown: Closing Messag
e Handler)
Nov 17 11:16:54 NOTICE[1996]: win32_tapi.c:244 telephonyShutdown: Closing lines.

Nov 17 11:16:54 NOTICE[1996]: win32_tapi.c:273 telephonyShutdown: Shutting down
TAPI.
Nov 17 11:16:54 ERROR[1996]: win32_tapi.c:276 telephonyShutdown: TAPI Error: 800
00050 on lineShutdown.
Nov 17 11:16:54 WARNING[1996]: loader.c:556 load_modules: Loading module chan_ta
pi.so failed!

Asterisk stopped

Statistics : Posted by aeldridge • on Mon Nov 17, 2014 10:38 am • Replies 3 • Views 78
↧

Trouble getting asterisk running on Server2012 R2 64 bit

November 17, 2014, 8:38 am
$
0
0
Seems to be trouble with the chan_tapi.so module. If you don't need that, then I'd axe it and see if the load continued without failing otherwise. If you do need it, then I'd find the people behind chan_tapi, I guess that's Patrick Deruel, for whom I have no contact information, or whomever's maintaining it these days, since it's not part of mainline Asterisk.

Statistics : Posted by aeldridge • on Mon Nov 17, 2014 10:38 am • Replies 3 • Views 78
↧
↧

Problem changing RTP Ports

October 17, 2014, 7:58 am
$
0
0
I solved it
The problem dissapeared when I change the value from free pbx and not from de conf file

Statistics : Posted by mmg_asterisk • on Fri Oct 17, 2014 7:58 am • Replies 1 • Views 237
↧

Problem changing RTP Ports

October 17, 2014, 7:58 am
$
0
0
Hello! Thanks in advance for reading

I've last asterisk now installed and I've a problem when I change RTP ports from rtp_additional.conf
I change, then I reload asterisk, and it works
The problem come when I change sth from free-pbx, and apply the changes. RTP ports change to the default values!!
Has anybody had the same problem? How can I solve it?

Thanks again!
Regars
Marco

Statistics : Posted by mmg_asterisk • on Fri Oct 17, 2014 7:58 am • Replies 1 • Views 237
↧

Comfort noise Support in 11.10.2?

November 17, 2014, 2:24 pm
$
0
0
Howdy,

Negative, Asterisk doesn't provide comfort noise.

Cheers

Statistics : Posted by medicranger • on Mon Nov 17, 2014 4:24 pm • Replies 1 • Views 29
↧
Viewing all 7460 articles
Browse latest View live
Search
<script src="https://jsc.adskeeper.com/r/s/rssing.com.1596347.js" async> </script>