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avaya tie trunk with Tiger Networks 3xxx ISDN card

August 27, 2014, 7:16 am
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With that card you should qualify for support from Digium. This forum does not provide support from Digium.

Statistics : Posted by jitbaaz • on Wed Aug 27, 2014 7:16 am • Replies 9 • Views 155
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avaya tie trunk with Tiger Networks 3xxx ISDN card

August 27, 2014, 7:16 am
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Mods,

this topic can be closed...I realized my error...didn't crimp by loopback cable hard enough...created another one, and voila. no alarms.

Kind Regards,
J

Statistics : Posted by jitbaaz • on Wed Aug 27, 2014 7:16 am • Replies 9 • Views 155
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avaya tie trunk with Tiger Networks 3xxx ISDN card

August 27, 2014, 7:16 am
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yup. I get red/yellow alarms with the loopback connector. without it, just red.

Statistics : Posted by jitbaaz • on Wed Aug 27, 2014 7:16 am • Replies 9 • Views 155
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avaya tie trunk with Tiger Networks 3xxx ISDN card

August 27, 2014, 7:16 am
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No,

Can also mean wrong configuration.

Did you try the loopback adaptor?

I would try that as well, after that, start checking your configu.

Good luck.

Statistics : Posted by jitbaaz • on Wed Aug 27, 2014 7:16 am • Replies 9 • Views 155
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avaya tie trunk with Tiger Networks 3xxx ISDN card

August 27, 2014, 7:16 am
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Hello Meightee,

Yup. Created that cable. After doing so, alarms in dahdi_scan changed from red -- red/yellow
I guess as per that kb though it would seem any alarm means a faulty card?

Kind Regards,
J

Statistics : Posted by jitbaaz • on Wed Aug 27, 2014 7:16 am • Replies 9 • Views 155
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avaya tie trunk with Tiger Networks 3xxx ISDN card

August 27, 2014, 7:16 am
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avaya tie trunk with Tiger Networks 3xxx ISDN card

August 27, 2014, 7:16 am
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Hello gurus,

I'm a newbie to building an asterix pbx box. So far I have installed Cent OS 6.5/Asterix 11/Dahdi.
My card that I am using is the single span Digium card, TE110P. Upon running dahdi_genconf and restarting dahdi and asterix services. Dahdi_scan shows an alarm of RED on the card. Is this normal? I haven't connected any voice cross over cable to the avaya ds1 card yet. Upon making a simple loop back plug, the alarm changes to RED/Yellow.
I'm curious to know if the RED alarm is normal for a fresh gen-configured card? All my reading so far states RED should go away once you plug in a loop back plug.

Kind Regards,
J

Statistics : Posted by jitbaaz • on Wed Aug 27, 2014 7:16 am • Replies 9 • Views 155
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lack of information in 'core show channels verbose'

August 28, 2014, 5:08 am
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We use the dialer software.
120 - dialer extension
630 - interviewer extension
600600600 - respondent number

Problem: no idea how to know what number the interviewer is speaking with, there is no correlation between 630 and 600600600

Question: should I modify dialplan somehow?

Below indicates the call in progress (dialer with SIP 120 called 600600600 number and transfer the call to SIP 630):
asterisk -rx 'core show channels verbose'
Channel Context Extension Prio State Application Data CallerID Duration Accountcode PeerAccount BridgedTo
SIP/120-0000fe1d macro-dial-one s 42 Up Dial SIP/630,,TtrI 120 00:51:18 SIP/630-0000fe1e
SIP/630-0000fe1e from-internal 1 Up AppDial (Outgoing Line) 630 00:51:18 SIP/120-0000fe1d
SIP/120-00010429 macro-dialout-trunk s 2 Up Dial DAHDI/g0/600600600,300,T 120 00:00:25 DAHDI/i1/600600600-15cb2
DAHDI/i1/600600600-15cb2 from-pstn 1 Up AppDial (Outgoing Line) 48600600600 00:00:25 SIP/120-00010429

Statistics : Posted by driverast • on Thu Aug 28, 2014 5:08 am • Replies 1 • Views 40
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lack of information in 'core show channels verbose'

August 28, 2014, 5:08 am
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core show channels is an engineering diagnostic; it is not a tool for doing CTI. Also, the summary level doesn't show the bridged channel, so you won't be able to put the call together (core show channel xxxx might, but I haven't checked.) If you want to do CTI use AMI.

Also note that the context names suggest you may be using FreePBX, and these forums are not suitable for questions about the intricacies of the FreePBX dialplan.

Statistics : Posted by driverast • on Thu Aug 28, 2014 5:08 am • Replies 1 • Views 40
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Retrieve_conf failed 255

August 28, 2014, 8:48 am
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I recently updated from 3.1 using the AsteriskNOW3.0.x-FreePBX5.211.65-13 script ... everything has been fine until I updated modules ... now when I "apply config" I get "Reload failed because retrieve_conf encountered an error: 255" ... here is the "more detail"
exit: 255
found language dir en_UK for broadcast, not installed on system, skipping
found language dir fr for directory, not installed on system, skipping
Added to globals: ASTETCDIR = /etc/asterisk
Added to globals: ASTMODDIR = /usr/lib/asterisk/modules
Added to globals: ASTVARLIBDIR = /var/lib/asterisk
Added to globals: ASTAGIDIR = /var/lib/asterisk/agi-bin
Added to globals: ASTSPOOLDIR = /var/spool/asterisk
Added to globals: ASTRUNDIR = /var/run/asterisk
Added to globals: ASTLOGDIR = /var/log/asterisk
Added to globals: CWINUSEBUSY = true
Added to globals: AMPMGRUSER = admin
Added to globals: AMPMGRPASS = amp111
Added to globals: AMPDBENGINE = mysql
Added to globals: AMPDBHOST = localhost
Added to globals: AMPDBNAME = asterisk
Added to globals: AMPDBUSER = freepbx
Added to globals: AMPDBPASS = fpbx
Added to globals: VMX_CONTEXT = from-internal
Added to globals: VMX_PRI = 1
Added to globals: VMX_TIMEDEST_CONTEXT =
Added to globals: VMX_TIMEDEST_EXT = dovm
Added to globals: VMX_TIMEDEST_PRI = 1
Added to globals: VMX_LOOPDEST_CONTEXT =
Added to globals: VMX_LOOPDEST_EXT = dovm
Added to globals: VMX_LOOPDEST_PRI = 1
Added to globals: MIXMON_DIR =
Added to globals: MIXMON_POST =
Added to globals: DIAL_OPTIONS = Ttr
Added to globals: TRUNK_OPTIONS = Tt
Added to globals: TRUNK_RING_TIMER = 300
Added to globals: MIXMON_FORMAT = wav
Added to globals: REC_POLICY = caller
Added to globals: RINGTIMER_DEFAULT = 25
Added to globals: TRANSFER_CONTEXT = from-internal-xfer
PHP Fatal error: Call to undefined function broadcast_backgrounddetect() in /var/www/html/admin/modules/broadcast/enc/functions.inc.php on line 73
1 error(s) occurred, you should view the notification log on the dashboard or main screen to check for more details.

I'm pretty new to Asterisk, so I don't even really know where to start ... I've done some google searching but haven't found much.

Any and all help would be appreciated.

Statistics : Posted by jerrygarcia4295 • on Thu Aug 28, 2014 8:48 am • Replies 0 • Views 8
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Cannot log in to Asterisk GUI: Asterisk Configuration Engine

April 8, 2007, 4:53 pm
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Clearing cookies did not work for me. Argh!

Statistics : Posted by mlanner • on Sun Apr 08, 2007 4:53 pm • Replies 9 • Views 18401
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Cannot log in to Asterisk GUI: Asterisk Configuration Engine

April 8, 2007, 4:53 pm
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mlanner wrote:Why can't I log in to the Asterisk Configuration Engine?




I had the same problem... To solve it I did clear cookies referred to server.

Statistics : Posted by mlanner • on Sun Apr 08, 2007 4:53 pm • Replies 9 • Views 18401
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Cannot log in to Asterisk GUI: Asterisk Configuration Engine

April 8, 2007, 4:53 pm
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Thx j4m3s.



I do have cmd line access. I think that's what I'll use to configure my SIP provider too. Although the GUI is nice, it appears that some of it isn't quite working yet. Seems like the cmd line is the way to go at the moment for many things.

Statistics : Posted by mlanner • on Sun Apr 08, 2007 4:53 pm • Replies 9 • Views 18401
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Cannot log in to Asterisk GUI: Asterisk Configuration Engine

April 8, 2007, 4:53 pm
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if you have command line access to the box, you can look at manager.conf and see what the password is to access the asterisk-gui.



Code: $ sudo vi /etc/asterisk/manager.conf




Underneath the '[admin]' section you'll see 'secret = xxxx' where xxxx is your pw.



The asterisk-gui, rPath system configuration gui, and linux user passwords are all different. Originally, if you install using a CD, the password is the same for the asterisk-gui and the 'admin' linux user.

Statistics : Posted by mlanner • on Sun Apr 08, 2007 4:53 pm • Replies 9 • Views 18401
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Cannot log in to Asterisk GUI: Asterisk Configuration Engine

April 8, 2007, 4:53 pm
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[SOLVED] Kind of ...



*********************



OK, so I reinstalled one more time, this time using ONLY lower case letters for the admin password DURING the setup. I can now, finally, get past the Configuration Engine page. (I guess changing the password after installation does not work?)



Now, I've got some other problems getting my inbound and outbound trunks to work, and the configs to stick when changing them from the GUI, but that's a different problem ...



*********************



Why can't I log in to the Asterisk Configuration Engine?



Install went smoothly. Base system config (from the "System Configuration" link in the upper right corner of GUI), including setting up IP address, updates, etc went fine. All updates have been downloaded and applied.



I've followed every tip and piece of advice I've found in this forum and in user documentation, but for some reason I keep on getting "Authentication failed" and I can't get past the Asterisk Configuration Engine login screen and start the setup.



I can log in to the System Configuration just fine. I know the root and the admin passwords, so that cannot be what's causing the problem logging into the Config Engine.



Originally I installed AsteriskNOW with a root p/w with a "secure" p/d. The original p/w was comprised of upper and lower case, and numbers. I read somewhere that apparently that was a no-no, so I changed it to a simple word all lower case letters. Argh ... no difference.



Some people seem to report that they can access the system from some computers but not others. Others say they "all of a sudden" were able to log in from a given machine although a day or two earlier they couldn't.



I've tried logging in from FF, Safari, IE ... from multiple machines, still no luck.



I also tried reinstalling AsteriskNOW. No luck with that either.



What am I missing here?



Please, please, someone ... I'm tearing my hair out ...



I have a Trixbox install, which works just "out of the box," but since I've read so much good about AsteriskNOW, I really want to try out the AsteriskNOW server also.



Thanks in advance to anyone who can give me some insight on this.

Statistics : Posted by mlanner • on Sun Apr 08, 2007 4:53 pm • Replies 9 • Views 18401
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Asterisk12 Webrtc Minimal installation

August 28, 2014, 1:26 pm
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I've been researching webrtc2sip and asterisk integration and am in the process now of building it on a seperate vm from my asterisk vm and i'll try to integrate the two products. I'm still not sure if this is necessary with asterisk 12 but most of the documentation i've found so far says that it was up until asterisk 11.x even with webrtc and ice support added.

can multiparty videoconferencing still be done through the asterisk if webrtc2sip is used?

I've installed asterisk in legacy polycom/cisco videoconferencing networks to tie the clients ip phones to their vtc systems by establishing a sip trunk from the pbx to the cisco gateway. would that type of setup still be possible with a third party application like webrtc2sip?

I hope these architecture questions are in the right place Image

Statistics : Posted by rcmiller • on Thu Aug 28, 2014 1:26 pm • Replies 1 • Views 68
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Asterisk12 Webrtc Minimal installation

August 28, 2014, 1:26 pm
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Good Day,

so far i've built a few small webrtc applications that mainly focused on browser to browser communication so nothing special was required. I am currently working on a project where the client has two important must-haves. The first is that they be able to communicate with their Polycom VTC systems, the second is that they be able to do multiparty calls.

With the current iteration of Asterisk and the added support of ICE, websockets and webrtc is anything necessary for an installation of this kind besides just Asterisk? will I need to use a SIP library (i.e. jssip) and Proxy (Kamailio or OpenSIPs) in order to communicate with the legacy devices. And lastly are there any examples of asterisk using an external transcoder? I'm assuming that transcoding is still necessary from the browser even though H.264 has been open sourced.

I don't need a step by step or anything, i'm just trying to find out what items are still required to make a browser-->legacy call work at this time.

Statistics : Posted by rcmiller • on Thu Aug 28, 2014 1:26 pm • Replies 1 • Views 68
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taskprocessor_push: tps is NULL!

August 28, 2014, 3:54 pm
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I cannot say that I've personally seen that error before.

You should be able to contact Digium technical support though for assistance with this.

Statistics : Posted by reolus • on Thu Aug 28, 2014 3:54 pm • Replies 1 • Views 56
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taskprocessor_push: tps is NULL!

August 28, 2014, 3:54 pm
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I currently have several asterisk servers up at various locations (several old and scheduled to be replaced.) We are getting this error daily on a new rollout and I am wondering if anyone else has come across it and if there is a suggested fix to correct it.

It ends up taking the place of the ast_msg_queue message traffic and is cleared by doing a core restart when convenient. We also tend to see the Unable to write to alert pipe on Message/ast_msg_queue error come up before it eventually becomes the "tps is NULL!"

(Example of what we don't see anymore after the error arrives)
Set("Message/ast_msg_queue", "MESSAGE(custom_data)=mark_all_outbound") in new stack
-- Executing [digium_phone_module@dpma_message_context:2] Set("Message/ast_msg_queue", "TMP_RESPONSE_URI=sip:10.87.1.178:5060") in new stack
-- Executing [digium_phone_module@dpma_message_context:3] Set("Message/ast_msg_queue", "MESSAGE_DATA(Request-URI)=sip:10.87.1.178:5060;ob") in new stack
-- Executing [digium_phone_module@dpma_message_context:4] Set("Message/ast_msg_queue", "MESSAGE_DATA(X-Digium-AppServer-Response-URI)=") in new stack
-- Executing [digium_phone_module@dpma_message_context:5] Set("Message/ast_msg_queue", "MESSAGE_DATA(X-Digium-AppServer-Response-FullContact)=") in new stack
-- Executing [digium_phone_module@dpma_message_context:6] MessageSend("Message/ast_msg_queue", "sip:10.87.1.178:5060,proxy") in new stack


Asterisk 12.4.0 loaded from Asterisk's rpm
DPMA
About 100 Digium D40, D70, and D50 phones using LLDP with firmware version=1_4_2_0_63880
Linux 2.6.32-431.23.3.el6.x86_64 #1 SMP Thu Jul 31 17:20:51 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux ((Centos 6.5))

Statistics : Posted by reolus • on Thu Aug 28, 2014 3:54 pm • Replies 1 • Views 56
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"match=" fails when using cdr format Asterisk 12

August 29, 2014, 10:58 am
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When using an "identify" section for inbound calls, Asterisk does not parse the address part when specified in cdr format as the help states.
Code: [Callcentric]
type=identify
endpoint=Callcentric
match = 204.11.192.0/24


returns no match: Endpoint: Callcentric Not in use 0 of inf
OutAuth: Callcentric/17771234567
Aor: Callcentric 0
Contact: Callcentric/sip:callcentric.com Unknown nan
Transport: transport_udp udp 0 0 192.nnn.nnn.nnn:5060



Statistics : Posted by proftech • on Fri Aug 29, 2014 10:58 am • Replies 0 • Views 6
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