Sorry to sound a bit silly, but I’ve tried to find the Blacklist feature in my installation of trixbox but cannot see it!
I’ve seen screenshots of FreePBX and it is there but its not in my list under PBX settings?
How can I install this feature?
I have a few cold callers I wish to block.
Cheers,
Bod.
Hi guys I have been trying to figure out why my inbound calls from one of my providers are arriving on my trixbox server running asterisk 1.4 but go no where.
I have an any any catch all inbound route for numbers the system can not match but for this I have an explesit inbound route I see the call arrive see below but it deos not go to its intented target IVR.
———————————————————————————–
Found the problem: In my trunk context I had from-internal I changed it to from-trunk and it fixed the problem.
We have been using Altigen PBX server (AltiServ 5.0.1.407) and its IP phone (IP600). Both the server and the IP phone only support H.323. For now, we got a new project to integrate the the Altigen box with a Trixbox CE server. The aim is to make it is able to call each office extension from both sides, and call the B city’s PSTN from the A city office, vice versa. The diagram should look like this: PSTN <-> A City office <- VOIP -> B City office <-> PSTN
As I am new to Trixbox (Asterisk), could you guys give me some suggestions on where I should start from? e.g.
1) How to make Trixbox CE support H.323 trunk?
2) What need to be modified in the Trixbox CE to make it work with Altigen?
…
Hi everyone,
Just a simple question:
Where or how do I download trixbox 2.8, - download link in the downloads section does not work.
any help? my box died few weeks ago and now I really need to reinstall it.
I found this on sourceforge, I hope this is the correct iso.
http://sourceforge.net/projects/asteriskathome/files/trixbox%20CE…
ok heres what i want it to do, I would like to set it up so taht when you dial in to the DISA,
first it asks your password,
then dial desired caller id,
then dial the number you are calling,
so basically i call in to the pbx, put in my password, dial my desired caller id, then the number i want to call, and it calls that number with the spoofed caller id. How could i go about doing this, i can do it on a web interface, but i cant seem to figure out how to do it on a dial in.
Hi, all!
I have set up a Trixbox (2.8) server, but it accept calls only sometimes. It may be a ok for a little while (a very few minutes), but for a while (probably mostly) there is something wrong.
In the log, all incoming calls starts like this:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
For calls that are accepted normally, the log continues normally. But if not, the logging ends here, and the caller finally gets a busy-signal (or something like that, probably generated by the voip-provider)…
Any ideas?
If it is of interest, the server is behind a firewall that has the ports udp 5060 and udp 10000-10100 forwarded to the server, rtp.sip changed accordingly, and externip/localnet is specified in sip_nat.conf. The firewall is a Juniper ssg5 (firmware Version: 6.1.0r3.0), with SIP and H323 disabled (as requested by the voip-provider).
Thanks in advance!
Frode
I have a handful of Polycom/Spectralink 8002 WiFi SIP phones that I have issues with receiveing calls. If the phones are power-cycled then they start receiving phone calls again. They can make phone calls as long as it isn’t another 8002. The deskphones never have this issue. They have been working like a champ.
Here is my sip_allusers.cfg file:
CODECS = g711u, g711a PROXY1_TYPE = Asterisk PROXY1_ADDR = 192.168.5.5:5060 # replace the ip address with the Asterisk Server's Address PROXY1_KEYPRESS_2833 = enable PROXY1_KEYPRESS_INFO = disable PROXY1_HOLD_IP0 = disable PROXY1_PRACK = enable PROXY1_REREG_SECS=300 PROXY1_KEEPALIVE_SECS=14 #PROXY1_DOMAIN = axlx.engr.local # Replace this with your SIP Domain's name PROXY1_CALLID_PER_LINE = disable PROXY1_MAIL_ACCESS = *97 # Put Your Voice Mail Sytem's Pilot Number here
Here is one my extension configs:
AUTH = 2401; f^Ey49_lZcvpH=o LINE1 = 2401 LINE1_PROXY = 192.168.5.5 LINE1_CALLID = Nursing Mobile
We seem to be getting nowhere with Fonality on how to make this work. Bear with me as I know next to nothing about routing tables and such. I have been through these forums and googled for a solution but can’t figure it out. Please tell me what I need to change and why it works that way. Here’s our setup:
Existing data network
Gateway 192.9.211.220
|
eth0 192.9.211.199
Fonality server
eth1 10.116.116.1
|
SIP phones (Polycom)
The phones are on their own subnet with their own POE switches. The phones can provision and talk to each other with no problems. However, if I put a phone on the eth0 side: it can call a eth1 phone (but no audio back), HUD doesn’t see the phone, it won’t provision, and eth1 phones can’t call the eth0 phone. This is what the route table looks like:
Kernel IP routing table Destination Gateway Genmask Flags Metric Ref Use Iface 1.128.6.1 * 255.255.255.255 UH 0 0 0 tun1 2.128.6.1 * 255.255.255.255 UH 0 0 0 tun0 192.9.211.0 * 255.255.255.0 U 0 0 0 eth0 192.168.219.0 1.128.6.1 255.255.255.0 UG 0 0 0 tun1 192.168.219.0 2.128.6.1 255.255.255.0 UG 0 0 0 tun0 10.116.116.0 * 255.255.252.0 U 0 0 0 eth1 169.254.0.0 * 255.255.0.0 U 0 0 0 eth1 default 192.9.211.220 0.0.0.0 UG 0 0 0 eth0
Hi,
I have 3 extensions (201, 202 & 203).
201 is X-Lite on my mac, 202 is siphon on my iPhone and 203 goes to a D-Link DVG-1120s which has a Philips voip433 connected to it.
When I call 201 -> 202, or 202 -> 201, or 203 -> 201/202 it all works. But when I want to call to extension 203 it doesn’t work. So I can call from 203 but not to 203.
Outbound calls go to an IVR, option 1 goes to call group 600 which is 201, 202 and 203. 201 and 202 ring, but not 203.
201, 202 and 203 have the same SIP settings. Even tried logging in as 201 using the D-link, but get the same problem but then for extension 201.
How can I solve this issue?
Best regards,
Paul Peelen
Hi, I’m kinda newb on trixbox and I’m stuck in a problem for days. I created an app on my company to find some info about the caller based on its number. Basically I send an http get:
http://callerid.intranet.tibra/retornacallerid.ashx?callerid=4196…
The website then returns the response in plain text:
D Paula Flores - cod 0 (Marli)
On trixbox, I set up the Caller ID Lookup Source like this:
Source description: http
Source type: HTTP
Cache results: no
Host: http://callerid.intranet.tibra
Port: 80
Path: /retornacallerid.ashx
Query: callerid=[NUMBER]
There’s no username or password set.
Then I related this source to my Inbound Routes. But still, when a call comes in, nothing happens, on sip debug no event raise concerning the cid lookup, looks like trixbox isn’t “trying” to get the cid like I configured.
My trixbox version is 2.6.
Am I missing something?
Hello,
I am just curius if this is normal Trixbox behaviour:
asterisk1*CLI> show channels
Channel Location State Application(Data)
Zap/1-1 s@from-pstn:1 Up Bridged Call(SIP/2137-b79169c0
SIP/2137-b79169c0 s@macro-dialout-trun Up Dial(ZAP/g0/18772344xxx|300|)
Zap/5-1 s@from-pstn:1 Up Bridged Call(SIP/2128-b720e208
SIP/2128-b720e208 s@macro-dialout-trun Up Dial(ZAP/g0/18669104xxx|300|)
4 active channels
2 active calls
Is it normal that if 2 calls is make 4 channels is occupying? I thoght one call = one channel. So if we use T1 = 23 channels only 11 calls can be made? It can’t be like that.
I assume I am wrong. Can someone explain?
Thank You,
MST
Hello, and thanks for the help!
I am trying to setup Trixbox to pass DTMF tones straight through to an analog extension port, without regenerating or touching the DTMF tones at all.
I have a an associate who tells me this is possible, but isn’t familiar enough with Trixbox specifically to tell me what file needs to be edited to make this happen.
(We are trying to use DID lines for an alarm receiver, and we think we have determined that Asterisk is fouling up the DTMF tones when regenerating them, making them unreadable to the receiver…)
Anyway, if anyone can tell me how to get Trixbox to pass DTMF tones straight through, let me know!
Thanks
I was browsing through Digium’s website when I noticed they have a Skype to Asterisk product available. http://www.digium.com/en/products/software/skypeforasterisk.php
Is this a feasible option for TrixBox? Or would it require extensive scripting or customizing? Please let me know, I would like to setup a Skype trunk without having to setup a seperate system (Siskyee, Uplink, etc).
Hello,
I belive I found a bug in the app_tx module but I do not know how to report it. When a fax does not go through because it called an incorrect phone number that turns out to be answered by a person or an answering machine instead of a fax machine, the result it passes back is passed when it should be failed. However, the module does indeed detect that it failed, just the result code is 0 instead of a failed indicator of a negative number. Here is a log snippet to demonstrate:
[Oct 27 08:40:51] DEBUG[28616] app_txfax.c: ==============================================================================
[Oct 27 08:40:51] DEBUG[28616] app_txfax.c: Fax send not successful - result (51) The call dropped prematurely.
[Oct 27 08:40:51] DEBUG[28616] app_txfax.c: ==============================================================================
[Oct 27 08:40:51] DEBUG[28616] app_txfax.c: execResult=0
How should I report this bug or what can I do to fix it?
Thanks!!
Ryan
