Re: [asterisk-users] Investigating international calls fraud
Did you have a look at the phone it self already? Is call forwarding activated or something and can you call the phone/extension from externally? I have seen this in the past where an employee enabled call forwarding on the phone and once at home he or family called the phone which forwarded the call to abroad. Good luck. Michel. Op 29-01-15 om 12:51 schreef d...@donkelly.biz: It's very unlikely that this was an employee calling Mom for 66 hours (I'm assuming these calls appeared on a single bill). It's also unlikely that someone inside would benefit financially from making these calls. (Follow the money!) Don't discount the possibility that you've overlooked something in the firewall. Meanwhile, does the client need to do international calling? If not, they may request that international calls be blocked by the carrier; once blocked, any international calls are the carrier's responsibility, not the client's. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Platt Sent: Thursday, January 29, 2015 12:11 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Investigating international calls fraud Hmm the calls are made during the day (and sometimes very early in the morning). Right now it looks like someone actually made these calls. If that is the case it's somewhat comforting to know the system wasn't compromised. However, the $25,000 phone bill still remains. Yikes. $6.25 per minute to Cambodia seems quite steep to me. Since the Mitel had a default admin password, it seems possible that somebody accessed its UI over the network, and then accessed and copied its SIP credentials for your Asterisk server. If that's the case, the calls might not have been placed through the phone. The miscreant could have configured the purloined credentials into another hardphone, or a softphone app on any PC or tablet or cellphone which was able to access your LAN. The cloned phone would not have needed to actually register with Asterisk... it could simply have send an INVITE to place a call, and Asterisk would have challenged it and then accepted the credentials. If your CDR log shows IP addresses for each call, you might be able to compare these with your DHCP (or whatever) IP registration service, and see if the calls actually came through the phone or not. If not you might be able to identify the device which initiated the calls. The bad news is, I suspect that you're probably on the hook for the cost of the calls. In the case of an inside job it's often hard to legitimately disavow the charges. You may have to pay the bill and then (if you can identify whomever placed the unauthorized calls) attempt to recover the cost from him/her in court. This sort of misused by an insider might be theft by conversion. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agi get_data noanswer
As we are top posting I will continue this.
Please have a look at:
https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
I hope this answers your questions.
Regards,
Michel.
op 13-08-14 01:34, Rafael Visser schreef:
I am talking about sip on asterisk 11.10.2
rv
2014-08-12 19:28 GMT-04:00 Eric Wieling ewiel...@nyigc.com
mailto:ewiel...@nyigc.com:
I do not know, maybe some of the other channel drivers sccp or sip
support it.
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Rafael Visser
*Sent:* Tuesday, August 12, 2014 7:24 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] agi get_data noanswer
Eric is correct. There is no way to send dtmf while the call has
not been answered.
But us very confusing the read command, in specific option =
n(noanswer) to read digits even if the line is not up
My AGI line is the following
$AGI-exec(READ,umenu,VARXX,1,n,2,7);
The command works, but there is no dtmf negotiation
$AGI-exec(READ,umenu,VARXX,1,,2,7);
The command works, but there is a kind of answer
What is the purpose of this noanswer option in a read command when
it is imposible to read?.
Is there any way to negotiate with the end user in this early
media situation?
Thanks in advance.
rv
2014-08-07 20:02 GMT-04:00 Eric Wieling ewiel...@nyigc.com
mailto:ewiel...@nyigc.com:
Generally the only thing you are allowed to do before answer is
send audio. You can't receive audio and can't receive DTMF. I
assume it is to prevent people from doing exactly what you are
trying to do --- trying to have two way communications without
paying for the call.
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Rafael Visser
*Sent:* Thursday, August 07, 2014 4:56 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] agi get_data noanswer
Hi Guys..
I am making an anoucement machine that is not allowed to answer
the call due to a billing issue.
I found that Playback with noanwser is usefull in this case.
$AGI-exec('Playback',$message,noanswer)}
But when i request some values to the user with get_data, i think
there is an answer anywere.
Is there a way to get_data without answering the call?
Thanks in advance!!
rv
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Re: [asterisk-users] Best approach in asterisk configuration
op 04-02-14 09:29, sylvain Gotri schreef: Hi , I have asterisk 1.8.5 installed on Centos 6. Now I want to configure my PBX to work in my network. I see that I can do this with asterisk files or use database like mysql to do it (realtime) I want to know what is the best way and what can be consequence when I choose other way ? Thanks. Silvain, My experience is that Realtime in mysql works good for sip users and queues. To put your dialplan into the database depends on how big it will be and how many calls it will handle. As the Asterisk documentation says when your dialplan grows the load on your database will grow exponentialy. This is because for each incoming call it will go through all the records in the dialplan. When your dialplan is in a text file this will only be loaded once on startup or when you do a manual reload. In my environment I have a static dialplan in the extensions.conf file and a dynamic part which i query through an AGI and php scripts. These scripts are optimized for querying the database. The dynamic dialplan in the database is managed by a custom made webservice (apache/php/mysql) Regards, Michel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS with a Portech MV-374 GSM Gateway
On 09-09-13 23:11, Niccolò Belli wrote: Hi, I have a Portech MV-374 GSM Gateway and I'd like to send SMS from a web page to confirm the subscriptions. How can I achieve it? Is Asterisk of any use to send SMS with the Portech? I really have no idea because I know nothing about the whole SMS thing... Thanks, Niccolò Niccolò, Reading the manual will help you. You do not need Asterisk at all to send SMS. You will need to use some scripting to use the API of the Portech device. Regards, Michel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need a second opinion on a new phone system deployment
Please also have a look at the gateway boxes from berofix (http://wiki.beronet.com/index.php/Main_Page). I am not affiliated but have used different products from them over last few yeas and all have survived and are stable. Documentation is open and free on their wiki. They provide updates. They are not the cheapest but they have different vendors and they are sold in online webshops. You can choose for the inside PCI(e) cards or their external boxes. Last few years I went for the external boxes. They can be fitted in a server rack or you mount them against the wall with screws. Regards, Michel. On 16-06-13 16:55, Nunya Biznatch wrote: Thanks again to everyone that's responded thus far. I have once again bundled the questions and answers into a single email, and am responding below. On 6/14/2013 9:43 AM, Nunya Biznatch wrote: Howdy All, They say opinions are like belly buttons, everybody has one. (that's the clean version of the saying). So I'm asking for yours. I hope you see it as a fun exercise. I'm designing a phone system from the ground up. Will be about 1000-1300 seats mixed 80/20 VoIP/Analog. 58-acre campus environment with 23 buildings. Userbase is emergency services organization, 24/7/365 operation. Down time is not an option, but blips are acceptable. Repair time is immediate. We need failover for the failover essentially. However, money is a major factor, so I have to do it all for nothing. So here's what I'm thinking. Please throw in your 2 cents. Network will be separate for phones. Fiber infrastructure available between buildings as well as copper. Internet access will be limited to a single administrative console on a temporary basis, and then only when remote 3rd party support is required. Access for 3rd party support will be supervised through remote access tools such as VNC, GoToMeeting, etc... etc... System will have zero access to local data network. This means all ancillary support servers such as DHCP, DNS, NTP, FTP, etc...etc... will be specific to the phone system. Yes, I know some responders at this time will become fixated on me gaining this connectivity. It ain't gonna happen. It's not an option. Period, end of story. These are the parameters I must work within. Trying to fix that will be a non-starter. The phone system will upgrade an existing TDM-based system. Mitel SX2000 with NuPoint Voicemail. This will not be a dump-trunk replacement. I expect at least a one to two-year transition, meaning we will have time to find problems, work bugs, and learn over time, with minimized impacts. It also means we'll be supporting two systems for some time. PBX is 97% serving your basic phone on the desk. Nothing special. Customers expect the usual list of features. There will be a goodly number of hints required for BLF on maybe 150 phones. There is one office of about 30 phones in a call-center environment that will need that service. They would be considered low volume (but don't tell them that). My Skills... I am not a Linux kung fu master, but I have built and managed my share of Linux servers on mutiple Linux flavors. I am a DCAA, having been through formal training, and have been playing with Asterisk for years, but always in fits and spurts and never in a live environment so I am by no means a kung fu master there either. I have started dabbling with virtualizations via XEN, but I am not comfortable enough with it to go live this first round. I can see myself implementing it in about three years once we're totally comfortable with what we have, so I can then have time to get that skill sorted. I was a network engineer for the US no3. telecom for a number of years, 10-years in comm-electronics in the military before that. Telecom my entire career. I've got the kung-fu to handle the network side of the house, and having administrated multiple PBXs for decade-plus, I've got the concepts down. No plans to build databases for things like directories, etc... I'm not greatly confident in those skills, and to date, haven't found anything that really stands out that would make me require that. You may think otherwise, so please chime in. I say that, but at the same time I recognize I may require a GUI interface once fully deployed to allow lower-skilled people to follow the motions to complete simple moves, adds, and changes. I'm fighting the uphill battle that is the GUI is new, CLI is old mentality. System will use G.722 for VoIP Phones. So there's the groundwork. Here's the hardware plan. Plan is to build my own servers following industry standards (ATX) and using industry standard equipment. Why? Spares? Whether redundant or not, I will still have spares for the most common elements on the shelf so equipment can be returned to service as quickly as possible. This will also allow me to be comfortable with more basic server configurations and help keep cost down. For example, Servers with single
Re: [asterisk-users] I need a second opinion on a new phone system deployment
Please also have a look at the gateway boxes from berofix (http://wiki.beronet.com/index.php/Main_Page). I am not affiliated but have used different products from them over last few yeas and all have survived and are stable. Documentation is open and free on their wiki. They provide updates. They are not the cheapest but they have different vendors and they are sold in online webshops. You can choose for the inside PCI(e) cards or their external boxes. Last few years I went for the external boxes. They can be fitted in a server rack or you mount them against the wall with screws. Regards, Michel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] monitoring Asterisk 1.8
We use Zabbix as monitoring tool and SNMP to get statistics and other info from Asterisk. for this you will have to make sure the snmp module for asterisk gets compiled and the Asterisk MIB is used. Regards, Michel. On 09-05-13 21:23, motty cruz wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, Hylafax and t38modem working together ?
Op 08-10-12 15:17, Olivier schreef: 2012/10/8 Michel Verbraak mic...@verbraak.org mailto:mic...@verbraak.org Op 08-10-12 09:24, Olivier schreef: Hi, I've read this thread in this list history http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match=t38modem http://sourceforge.net/tracker/?func=detailaid=3337581group_id=152230atid=783657 Has anyone been successful when integrating latest version of Asterisk (10 or 1.8, for instance) with t38modem ? My target setup is: fax ---PSTN-- SPA3102 ---T38-- Asterisk with t38modem and Hylafax Suggestions ? Yup, YaJHFC http://www.yajhfc.de/ --- Hylafax --- t38modem --- BeroFix http://www.beronet.com/product/berofix-gateways/ --- ISDN32 By the way, which t38modem did you use ? On my debian system, version 1.2 is packaged and I wonder if it's worth the effort to use lastest 2.0 version. We use the 1.2.0-1 version on a debian system. No Asterisk in this case but it does work excelent. With the YaJHFC software you get a Windows/Linux/OSX printer driver. The BeroFix could be replaced with Asterisk but I do not have tested this. Regards, Michel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, Hylafax and t38modem working together ?
Op 08-10-12 09:24, Olivier schreef: Hi, I've read this thread in this list history http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match=t38modem http://sourceforge.net/tracker/?func=detailaid=3337581group_id=152230atid=783657 Has anyone been successful when integrating latest version of Asterisk (10 or 1.8, for instance) with t38modem ? My target setup is: fax ---PSTN-- SPA3102 ---T38-- Asterisk with t38modem and Hylafax Suggestions ? Yup, YaJHFC http://www.yajhfc.de/ --- Hylafax --- t38modem --- BeroFix http://www.beronet.com/product/berofix-gateways/ --- ISDN32 No Asterisk in this case but it does work excelent. With the YaJHFC software you get a Windows/Linux/OSX printer driver. The BeroFix could be replaced with Asterisk but I do not have tested this. Regards, Michel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Termination Provider Madness
Op 03-10-12 01:17, Chris Nighswonger schreef: On Tue, Oct 2, 2012 at 5:30 PM, Chris Bagnall aster...@lists.minotaur.cc wrote: On 2/10/12 6:51 pm, Carlos Alvarez wrote: Your traffic level, number of concurrent calls, etc would help us know what sort of carrier you should be talking to. Equally important, your geographic location, and the geographic location to which most of your calls are made will be useful in helping list members advise you. We do ~4000+ min of outbound calling per month and just about that inbound. Not a large volume. We have four DID's (one of which is 800). Our calling patterns are mostly the lower 48 with a smattering international. We are located in NC. RTP is the problem in the FW. I just cannot see opening all RTP ports to $universal. But I'm probably missing a key piece of information. :-) Kind Regards, Chris Chris, Have a look at your /etc/asterisk/rtp.conf file. In it you specify the UDP portrange your asterisk will use for RTP traffic. change the rtpstart and rtpend to your needs and set them open in your FW. Do not make the range too small each active call will normally take one RTP channel incoming and one RTP channel outgoing. I have mine set to for example: rtpstart=1 and rtpend=10100. This should be enough for 100 simultanious calls. Regards, Michel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Termination Provider Madness
Op 03-10-12 15:08, Tim Nelson schreef: - Original Message - Have a look at your /etc/asterisk/rtp.conf file. In it you specify the UDP portrange your asterisk will use for RTP traffic. change the rtpstart and rtpend to your needs and set them open in your FW. Do not make the range too small each active call will normally take one RTP channel incoming and one RTP channel outgoing. I have mine set to for example: rtpstart=1 and rtpend=10100. This should be enough for 100 simultanious calls. 2 RTP ports per session (inbound/outbound media)... that would mean 50 simultaneous calls, no? --Tim -- Tim, As Far as I known are the outbound RTP ports determined by the other end. It is also UDP traffic so the inbound stream could be destined for port 1 and the outbound could be coming from port 1. So still 100 simultanious calls. 1 -- XXX (outbound) 1 - XXX (inbound) for one call. Michel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE110P Wildcard does not work with Ubuntu 12.04 server
Op 22-08-12 12:09, Shitian Long schreef: I am trying to setup TE110P wildcard on a PBX running ubuntu 12.04 server edition. I followed the procedure from http://docs.digium.com/misc/ADL_quickstart.pdf step by step. During the process of installing dahdi-linux-complete I got following warnings: root@ubuntu:/usr/local/src/dahdi-linux-complete-2.6.1+2.6.1# make perl: warning: Setting locale failed. perl: warning: Please check that your locale settings: LANGUAGE = en_US:en, LC_ALL = (unset), LC_CTYPE = UTF-8, LANG = en_US.UTF-8 are supported and installed on your system. perl: warning: Falling back to the standard locale (C). Frist of, I am wondering if this error matters? Second question, after installation process complete, and reboot the machine I got the following error, when machine boot up: Loading DAHDI hardware modules: wcte11xp: error I think the TE110P card is no properly loaded. I try to confirm my thought by using root@ubuntu:~# dahdi_tool There is no interface listed on the table. I am wondering if anyone got idea about this issue. Thanks. longst -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please try the command lspci and see if the card is mentioned in the results. Regards. Michel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calendar Integration Problem
On 30-04-12 11:09, Bharat Lalcheta wrote:
Hiii all,
I am using asterisk 1.8.9.2 and compile all modules related to calendar.
neon version is 0.29.6. OS is ubuntu 11.10.
I configured ical for zimbra, caldav for google mail and ews for
exchange 2010 calendar.
ical and caldav setup working fine and i am getting my calendar events
perfectly. But for exchange 2010 calendar i am getting following error.
Unable to communicate with Exchange Web Service at
'https://ex1.domain.com/EWS/Exchange.asmx': Could not authenticate to
server: ignored NTLM challenge, GSSAPI authentication error:
Unspecified GSS failure. Minor code may provide more information:
Credentials cache file '/tmp/krb5cc_0' not found
my calendar.conf is as follows
[calendar3]
type = ews ; type of calendar--currently supported:
ical, caldav, exchange, or ews
url = https://ex1.domain.com/EWS/Exchange.asmx ; URL to MS Exchange EWS
user = myn...@domain.com mailto:myn...@domain.com ;
Exchange username
secret = xx ; Exchange password
refresh = 10 ; refresh calendar every n minutes
timeframe = 20
calendar show status command shows following output
Calendar Type Status
--
calendar3ewsfree
Please help me out for solve above problem.
Thanks in advance
--
Bharat Lalcheta
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Did you compile neon against openssl or the default internal ssl? I used
against openssl.
Make sure you have the rootca from the exchange server in /etc/ssl/certs
The message/warning looks like the Exchange server expects a kerberos
authentication. I have no experience with the EWS calendar module and
using kerberos to authenticate.
Hope this info helps.
Michel.
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Re: [asterisk-users] No extension found ?
On 21-04-12 08:19, Olivier CALVANO wrote:
Hi
I have a small problems with incoming call.
I have a peer actually configured for outcall :
sip.conf:
[Trunk-Telco]
type=peer
host=domaineofmysupplier.net
outboundproxy=domaineofmysupplier.net
session-timers=originate
session-expires=7200
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
dtmfmode=rfc2833
disallow=all
allow=alaw
insecure=port,invite
context=incoming
This SIP account work for outgoing call. when i want receive call from
this sipplier, i have a extension not found.
In extensions.conf for incoming:
[incoming]
exten = _X.,1,Dial(IAX2/VoIP/${EXTEN},180,rt)
in dialplan show incoming, no problems i see the dialplan.
when i call, i have:
--- SIP read from UDP://84.xx.xx.72:5060 ---
INVITE sip:331NUMNOFOUND@78.IPOFMYSERVER:5060 SIP/2.0
Record-Route:sip:84.xx.xx.72;r2=on;lr;f=4
Record-Route:sip:172.16.21.172;r2=on;lr;f=4
Record-Route:sip:172.16.21.67;lr;f=8
Record-Route:sip:172.16.20.119;lr;did=247.29f60367
Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0
Via: SIP/2.0/UDP 172.16.21.67;branch=z9hG4bK10e4.bbf4c444.0
Via: SIP/2.0/UDP 172.16.20.119;branch=z9hG4bK10e4.9fe53c91.0
Via: SIP/2.0/UDP 172.16.21.11:5060;branch=z9hG4bK00151747E2606DB6CA39464AF542
From: +331MYCLID
sip:+331MYCLID;tgrp=RT43@172.16.21.11;tag=2RUVP51HBW3E1D1u0K4NFQC0QNAN31
To:sip:+331NUMNOFOUND@172.16.20.119
Call-ID: 60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
CSeq: 20114 INVITE
Contact:sip:+331MYCLID@172.16.21.11:5060
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
Content-Type: application/sdp
Max-Forwards: 67
P-Asserted-Identity:sip:+331myc...@domaineofmysupplier.net
Supported: timer, replaces
Content-Length: 369
Min-SE: 90
Session-Expires: 300
P-Charging-Vector: icid-value=4f924d2c1e20abe1d@172.16.20.119
X-PSN-Trunk: ME
v=0
o=- 18406958643964291255 1 IN IP4 172.16.21.11
s=session
c=IN IP4 84.xx.xx.34
t=0 0
m=audio 64296 RTP/AVP 8 18 4 0 105 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=fmtp:4 bitrate=6.3
a=rtpmap:0 PCMU/8000
a=rtpmap:105 X-CCD/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
a=nortpproxy:yes
-
--- (25 headers 17 lines) ---
== Using SIP RTP CoS mark 5
Sending to 84.xx.xx.72 : 5060 (no NAT)
Using INVITE request as basis request -
60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
No matching peer for '+331MYCLID' from '84.xx.xx.72:5060'
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 101
Peer audio RTP is at port 84.xx.xx.34:64296
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format PCMU for ID 0
Found unknown media description format X-CCD for ID 105
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined
- 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 84.xx.xx.34:64296
Looking for 331NUMNOFOUND in default (domain 78.IPOFMYSERVER)
It is looking for the 331NUMNOFOUND in context named default.
Do you have this context? Does the extension exists in the context?
Do you have a register line in your sip.conf for this external provider?
In the register line you can specify the extensions/device to use in the
sip.conf so it knows the right context to start in extensions.conf
instead of the default context.
For example: register = username:passw...@sip.voipbuster.com/Trunk-Telco
--- Reliably Transmitting (no NAT) to 84.xx.xx.72:5060 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0;received=84.xx.xx.72
snip
[Apr 21 08:01:16] NOTICE[11906]: chan_sip.c:18527
handle_request_invite: Call from '' to extension '331NUMNOFOUND'
rejected because extension not found.
Regards,
Michel.
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Re: [asterisk-users] Dialing multiple endpoints and CallerID presentation
Hi,
Use the local channel
Dial(Local/@contextinternallocal/b@contextexternal)
In the internal context you set CALLERID(num) to the internal extension
and then dial the SIP
exten = ,1,Set(CALLERDI(num)=${EXTEN})
same = n,Dial(SIP/${EXTEN})
In the external context do almost the same but dial DAHDI
exten = bb,1,Set(CALLERDI(num)=051234)
same = n,Dial(DAHDI/g1/0123456789)
Regards,
Michel.
Op 29-08-11 09:15, Olivier schreef:
Hi,
I've got the following use case where I want to simultaneously dial 2
endpoints that both need different CallerID presentation.
How can I do that, from the dialplan preferably ?
For instance, let say phone A needs to both dial B, an internal SIP
phone and C, a cell phone reachable through a DAHDI span from a an
Asterisk system where :
1. users can use 4-digits short numbers to reach other internal phones.
2. calls going out through the DAHDI span, must have CallerIDs
presented without any prefix.
Ideally, CallerID should be presented :
1- with 4-digits for internal phones
2- with 10-digits for external phones
so that both phones can return the call without re-dialing.
Suggestions ?
A is 1234 alias DID 051234
B is 5678
C is 0123456789
I was thinking of using something like this:
Dial(SIP/5678option_to_present_1234_to_calleeDAHDI/g1option_to_present_051234/0123456789)
What could be option_to_present_1234_to_callee and
option_to_present_051234
Regards
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Re: [asterisk-users] Request: please test modification to EWS calendar functionality
Op 12-06-11 15:38, Anders Fudali schreef:
Hi again,
In my environment, I have my phones configured with the same username
as the ActiveDirectory login and in order to map an incoming call to a
username, I simply do a SQL query against a user and phone
provisioning system that I've built.
Therefor selecting the right user is easy in my case, I guess some
sort of extension function would suffice for me. But I suppose that a
better way would be to collect and store all user specific settings
in a common place, such as in the sip configuration file.
If possible, why not add both methods? Then you would probably have
covered everyones needs.
Regards
Anders Fudali
From: Michel Verbraak mic...@verbraak.org mailto:mic...@verbraak.org
Date: Sun, 12 Jun 2011 14:59:02 +0200
To: Anders Fudali anders.fud...@jajja.com
mailto:anders.fud...@jajja.com
Subject: Re: [asterisk-users] Request: please test modification to EWS
calendar functionality
Op 12-06-11 13:05, Anders Fudali schreef:
Hi Michel,
I have a question regarding your recent added patch to the EWS
calendar function.
Would it be possible from the dialplan to specify which users
calendar that I'd like to query? I'm looking for a way to query my
users calendars on incoming calls for out of office or busy
events without having to specify each user in the calendar
configuration file.
Let me know if this is possible, thanks in advance.
Best regards
Anders Fudali
Anders,
Currently this is not possible. I have the same question on my to-do
list from my Boss but it has low priority for now.
For your question: How would you select the right user calendar in the
dialplan? Do you want an extension - exchangeuser table? Or extra
field in sip accounts?
Regards,
Michel
Anders,
Please do not top post for readability. I also included the list again
because the previous post was send to me directly.
I was thinking of creating a new function call, from the dialplan, where
you can specify a calendar from the calendar.conf file for the
url+user+secret and as extra arguments the two new fields folderbase and
folderpath.
Something like
Set(status=${GET_CURRENT_CALENDAR_STATUS(calname,folderbase,folderpath)})
This extension to the calendar module will be a seperate patch and for
that to work the current one on the reviewboard needs to be included
into the main trunk of asterisk.
Are you able to test the current upload on the reviewboard and post your
results back to the board?
Regards,
Michel.
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[asterisk-users] Request: please test modification to EWS calendar functionality
I have expanded the EWS calendar functionality within Asterisk 1.8 so it is now possible to access any calendar within an Exchange 2007 or 2010 server. I have put the changes onto the reviewboard for astrisk but currently no one responded. So if you use the EWS calendar functionality within Asterisk and would like to have access to any calendar in Exchange please try the patch in the following review request: https://reviewboard.asterisk.org/r/1152/. Please reply to the reviewboard if it is working for you or if you experience problems. Regards, Michel Verbraak. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] standalone PRI-to-SIP converter
Op 27-05-11 17:10, Michelle Dupuis schreef: I'm looking for recommendations for standalond PRI to SIP converters. (Needs to be outside the asterisk box - so a PCIe card won't do) I've used redfone but this project doesn't need the redundancy features... Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have a look at the berofix boxes from beronet (http://www.beronet.com/?page_id=358preview=true) Regards, Michel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maniuplate callerID based off of callerID
Almost,
If you use Asterisk version 1.6 or higher use
Exten = ExecIf($[${CALLERID(number)} = 400]?Set(CALLERID(num)=)
Or
Exten = ExecIf($[${CALLERID(number)} = 400]?Set(CALLERID(all)=)
Michel Verbraak
**
http://www.intercommit.nl/
On 08-04-11 15:56, Louis Carreiro wrote:
Hey all!
I'm trying to figure out a way to manipulate a call's caller ID based
off of the caller's caller ID. Basically, I've got a situation where
anything that goes through an Nortel Opt11's IVR comes out with the
caller ID 400 (the Opt11's IVR's ext). When the call goes out the
trunk that the call is destined for, I'd like to grab the 400 caller
ID and delete it so it comes through as Unknown. The Unknown part
doesn't have to be the literal string, just a blank CallerID would be
fine.
Would it be something like:
Exten = ExecIf($[${CALLERID(number)} = 400]?SetCallerID())
Thanks all!
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Re: [asterisk-users] Asterisk 1.8.3
We also see the random freeze of asterisk 1.8.3.2. We do use realtime. I have just applied the patch and will see how our environment holds. I will report back to the issue mentioned by Ishfaq Michel Verbraak *InterCommIT bv* ** On 06-04-11 09:44, Ishfaq Malik wrote: On Tue, 2011-04-05 at 21:10 -0400, Bryant Zimmerman wrote: I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. Thanks Bryant -- Could it be this issue? https://issues.asterisk.org/view.php?id=18818 Mind you, this one will only affect you if you use RealTime architecture -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys 962
Jeff LaCoursiere schreef: Working with a new client that has a ton of these phones, and in a new installation the phone is registered, can place and receive calls with no issues, but has a locked picture of a phone in the upper right corner. Any Linksys experts know what this means? I have searched the admin guide and googled to no results... really just an annoyance I suppose, but I would like to know what it means :) Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, We have a bunch of 922 and 962 phones. On the 922 phone a phone and a lock means that the phone is locked by a password for requesting possible sensitive information, last numbers dialed - phone settings - etc.., through the use of the phone's buttons. For example the recently dialed list of phone numbers, one of the buttons under the display, can only be seen after you enter a password. You press the button and the phone asks for a password. Changing of the the phone settings, ip - sip proxy - etc.., is also protected by password. We have our phones in a provisioning network where we manage which phones are locked for access to sensitive information and which are not. Regards, Michel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 w/ TE420B EC
trebaum schreef: I keep getting a red alarm when trying to setup asterisk to use my TE420B EC. I only have a blank context setup in my extensions.conf as I haven't started to config that until I can clear this red alarm. I don't have physical access to the server, so I can't go reseat the modules/card/ethernet cable, though I have hands on location that have done this a couple times already. Please help. I'm quite frustrated at this point. Thank you in advance for any help. */etc/dahdi/system.conf* # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Global data loadzone= nl defaultzone = nl */etc/asterisk/chan_dahdi.conf* [trunkgroups] [channels] ; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) group=1 context=frompstn switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31 context = default *cat /proc/dahdi/1* Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 RED 1 TE4/0/1/1 Clear (In use) RED(SWEC: MG2) 2 TE4/0/1/2 Clear (In use) RED(SWEC: MG2) 3 TE4/0/1/3 Clear (In use) RED(SWEC: MG2) 4 TE4/0/1/4 Clear (In use) RED(SWEC: MG2) 5 TE4/0/1/5 Clear (In use) RED(SWEC: MG2) 6 TE4/0/1/6 Clear (In use) RED(SWEC: MG2) 7 TE4/0/1/7 Clear (In use) RED(SWEC: MG2) 8 TE4/0/1/8 Clear (In use) RED(SWEC: MG2) 9 TE4/0/1/9 Clear (In use) RED(SWEC: MG2) 10 TE4/0/1/10 Clear (In use) RED(SWEC: MG2) 11 TE4/0/1/11 Clear (In use) RED(SWEC: MG2) 12 TE4/0/1/12 Clear (In use) RED(SWEC: MG2) 13 TE4/0/1/13 Clear (In use) RED(SWEC: MG2) 14 TE4/0/1/14 Clear (In use) RED(SWEC: MG2) 15 TE4/0/1/15 Clear (In use) RED(SWEC: MG2) 16 TE4/0/1/16 HDLCFCS (In use) RED 17 TE4/0/1/17 Clear (In use) RED(SWEC: MG2) 18 TE4/0/1/18 Clear (In use) RED(SWEC: MG2) 19 TE4/0/1/19 Clear (In use) RED(SWEC: MG2) 20 TE4/0/1/20 Clear (In use) RED(SWEC: MG2) 21 TE4/0/1/21 Clear (In use) RED(SWEC: MG2) 22 TE4/0/1/22 Clear (In use) RED(SWEC: MG2) 23 TE4/0/1/23 Clear (In use) RED(SWEC: MG2) 24 TE4/0/1/24 Clear (In use) RED(SWEC: MG2) 25 TE4/0/1/25 Clear (In use) RED(SWEC: MG2) 26 TE4/0/1/26 Clear (In use) RED(SWEC: MG2) 27 TE4/0/1/27 Clear (In use) RED(SWEC: MG2) 28 TE4/0/1/28 Clear (In use) RED(SWEC: MG2) 29 TE4/0/1/29 Clear (In use) RED(SWEC: MG2) 30 TE4/0/1/30 Clear (In use) RED(SWEC: MG2) 31 TE4/0/1/31 Clear (In use) RED(SWEC: MG2) ~T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users As I see you have specified nl as defaultzone so I expect that you are using a ISDN-30/15 line from provider KPN in the Netherlands. If so then remove the crc4 option from the span line in /etc/dahdi/system.conf. */etc/dahdi/system.conf* # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Global data loadzone= nl defaultzone = nl KPN is not using the crc4 checksum and therefore the card is not getting the wrong checksum on the lines and so they get a red alarm status. After the change reload dahdi and your lines should change colours. If this is working for you please answer to the mailing list so people in the future will find it. The next time please specify the type of telephoneline and provider. Regards, Michel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to detect switch to voicemail when calling to mobile phone
Hello, First of all I have an Asterisk setup of Asterisk 1.6.0.9 + DAHDI 2.0 + E1 card with ISDN-15 line (KPN Netherlands). I have two questions/situations: A. I would like to be able to interrupt the dial command when I try to call to a mobile phone and this phone is never answered by a person but after a certain time switches to the voicemail of the mobile phone. B. I have a setup where someone calls one extension in my asterisk which in turn will call about 6 mobile phones at the same time and the first person that picks up his mobile phone will get the caller. The 6 mobile phones need to get called until a real person answers one of the phones. We run into the problem when one of the called mobile phones switches to voicemail before any one of the 6 persons answers his phone. Is there a way that the call to the group ignores the mobile phones which switch to voicemail but keeps calling the others. When I turn on logging for dahdi or sip I can see that when the connection is switched to the voicemail a progress message is created (twice). This only happens when a switch to voicemail is made. Following is an extraction of the dahdi log when voicemail kicks in, see bold lines: Message type: SETUP (5) Message type: STATUS (125) Message type: CALL PROCEEDING (2) Message type: ALERTING (1) * Message type: PROGRESS (3) Message type: PROGRESS (3)* Message type: CONNECT (7) Message type: CONNECT ACKNOWLEDGE (15) Message type: DISCONNECT (69) Message type: RELEASE (77) Message type: RELEASE COMPLETE (90) Next is an extraction of the dahdi log when the mobile phone is answered by a person: Message type: SETUP (5) Message type: STATUS (125) Message type: CALL PROCEEDING (2) Message type: ALERTING (1) Message type: CONNECT (7) Message type: CONNECT ACKNOWLEDGE (15) Message type: DISCONNECT (69) Message type: RELEASE (77) Message type: RELEASE COMPLETE (90) Is there an option for the dial command to stop the call when the switch is detected and tell the caller that voicemail is active and if he would like to leave a message or not? Can I create/detect this with an AGI script and act on it? (what to look for). Any help is appreciated. Regards, Michel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HP DL380 G5 with TE420
Hans Konings schreef: Hi I'm having problems getting the TE420 working in HP DL380G5 servers. The cards don't seem to be detected 100% by the BIOS. With two cards in the server they are never detected. things I've tried: 1 Update firmware to latest (P56) for the server 2 change irq settings 3 disable all onboard devices on server and remove raid controller 4 different cards in different slots What I mean by not detected is that in the HP utility SMBIOS you can read out the status of the pci-express slot and it says slot available also lspci does not list the card. Every so often the card is detected and works properly. I've tried with a different server (HPdl320g5p) and the card is detected in this but the cards generate NMI errors on many bootups. Does anybody have this combination of hardware working? Or can anybody think of something I've missed? Rgds Hans I have a HP DL380G5 (dual quadcore) with a TE121 (PCI-E) card which works like a charm. Regards, Michel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE121 on Asterisk
Oguzhan Kayhan schreef: Oguzhan Kayhan wrote: I want to change it to E1 instead of T1. here comes the problem. If it's anything like the older cards, there is a jumper on the card that sets it to T1/E1 Doug Yes, I just noticed the jumper on the card. Thanks a lot. Yes i changed the jumper to enable E1. dahdi_scan shows the following [1] active=yes alarms=UNCONFIGURED description=Wildcard TE121 Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE121 with VPMADT032 location=PCI Bus 04 Slot 09 basechan=1 totchans=31 irq=17 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding= framing= But when i try to run dahdi_genconf i got the following error. 31 channels in a T1 span at /usr/local/share/perl/5.10.0/Dahdi/Span.pm line 244. dahdi_hardware 31 channels in a T1 span at /usr/local/share/perl/5.10.0/Dahdi/Span.pm line 244. There is a bug in the perl module. I had the same problem. It trips over the following part *WCT1/0* in *WCT1/0* Wildcard TE121 Card 0 (MASTER) HDB3/CCS. It finds the text T1 in there and expects it to be a T1 jumpered card in stead of an E1 jumpered card and it tries to create a T1 system.conf file. I still need to make a bug report about this. Your card is probably working allright. Create the right system.conf file in /etc/dahdi/ Mine has (E1 for Dutch KPN ISDN15/20/30) and the following lines: /# Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) HDB3/CCS/CRC4 RECOVERING span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 / Probably you have to enter ,CRC4 at the end of the span line. (span=1,1,0,ccs,hdb3,crc4) When you edited the file do a: # dahdi_cfg - DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) SNIP Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17) SNIP Channel 31: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 31) 31 channels to configure. Setting echocan for channel 1 to mg2 SNIP And followed by: #cat /proc/dahdi/1 Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) HDB3/CCS IRQ misses: 1 1 WCT1/0/1 Clear (In use) (EC: MG2) 2 WCT1/0/2 Clear (In use) (EC: MG2) 3 WCT1/0/3 Clear (In use) (EC: MG2) 4 WCT1/0/4 Clear (In use) (EC: MG2) 5 WCT1/0/5 Clear (In use) (EC: MG2) 6 WCT1/0/6 Clear (In use) (EC: MG2) 7 WCT1/0/7 Clear (In use) (EC: MG2) 8 WCT1/0/8 Clear (In use) (EC: MG2) 9 WCT1/0/9 Clear (In use) (EC: MG2) 10 WCT1/0/10 Clear (In use) (EC: MG2) 11 WCT1/0/11 Clear (In use) (EC: MG2) 12 WCT1/0/12 Clear (In use) (EC: MG2) 13 WCT1/0/13 Clear (In use) (EC: MG2) 14 WCT1/0/14 Clear (In use) (EC: MG2) 15 WCT1/0/15 Clear (In use) (EC: MG2) 16 WCT1/0/16 HDLCFCS (In use) 17 WCT1/0/17 Clear (EC: MG2) 18 WCT1/0/18 Clear (EC: MG2) 19 WCT1/0/19 Clear (EC: MG2) 20 WCT1/0/20 Clear (EC: MG2) 21 WCT1/0/21 Clear (EC: MG2) 22 WCT1/0/22 Clear (EC: MG2) 23 WCT1/0/23 Clear (EC: MG2) 24 WCT1/0/24 Clear (EC: MG2) 25 WT1/0/25 Clear (EC: MG2) 26 WCT1/0/26 Clear (EC: MG2) 27 WCT1/0/27 Clear (EC: MG2) 28 WCT1/0/28 Clear (EC: MG2) 29 WCT1/0/29 Clear (EC: MG2) 30 WCT1/0/30 Clear (EC: MG2) 31 WCT1/0/31 Clear (EC: MG2) I have a ISDN15 connected to it so only 15 lines are in use. If you see something like YELLOW or RED or BLUE in the previous something is wrong with your line. I had this first but this was because I had the CRC4 option added to my system.conf file. Your syslog log file /var/log/messages wil tell also if you have an alarm. Regards, Michel. What should i do about it? -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
