Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone
I have not use a TDM4xx card for a while, but I remember that in order for ringing to work, you had to plug in an extra molex connector into the card to supply power to the ringing generator portion. If you forgot to do that... Lyle BTW, I know about being a noobie. I was there once myself and still am there every day learning and working with new stuff. Sometimes not of my own choosing, but one must do what they need to keep getting those paychecksGRIN! On 6/20/2012 8:44 AM, Joseph Towery wrote: Thanks Lyle, Sorry to sound so much like a newb but in asterisk I am. I was initially trying to do things by hand in the extensions.conf file and had no luck. I then got from SVN checkout asterisk-gui and used it to simply try and get things started, and created a trunk, users, incoming rule, etc. from the gui and finally got dial tone, and can dial out, but I haven't got the analog phone ringing yet. I will have more targeted questions in the near future. It is just hard to find google help for analog answers. Most deal with SIP (which is my next step once I have the analog lines working). Thanks, *From:* Lyle Giese l...@lcrcomputer.net *To:* asterisk-users@lists.digium.com *Sent:* Tue, June 19, 2012 9:29:12 PM *Subject:* Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone An FXO port needs to be connected to dial tone or your PSTN line. And an FXS port needs to be connected to the station equipment(ie. a physical phone). The TDM410 is basically a channel bank to Asterisk, so the channel type inside Asterisk is FXO to talk to the physical FXS card and FXS to talk to the physical FXO port. Lyle Giese LCR Computer Services, Inc. On 06/18/12 15:08, Joseph Towery wrote: Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk samples so I realize that may have messed me up. This is all running on Ubuntu Server 12.04. I have been googling/researching reading the book, etc. Everything I find is for SIP softphones etc. I just want to start by getting the asterisk machine to provide dialtone to the analog phone, and ring that phone when I call the PTSN line. I must be missing something in the basic dahdi and dialplan to simple get the analog phone to work. Can someone point me to a example of what I am trying to accomplish? Not wanting handholding but a push in the right direction. Thanks. -- _ -- Bandwidth and Colocation Provided byhttp://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] TDM410 PTSN line setup with 1 analog phone
An FXO port needs to be connected to dial tone or your PSTN line. And an FXS port needs to be connected to the station equipment(ie. a physical phone). The TDM410 is basically a channel bank to Asterisk, so the channel type inside Asterisk is FXO to talk to the physical FXS card and FXS to talk to the physical FXO port. Lyle Giese LCR Computer Services, Inc. On 06/18/12 15:08, Joseph Towery wrote: Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk samples so I realize that may have messed me up. This is all running on Ubuntu Server 12.04. I have been googling/researching reading the book, etc. Everything I find is for SIP softphones etc. I just want to start by getting the asterisk machine to provide dialtone to the analog phone, and ring that phone when I call the PTSN line. I must be missing something in the basic dahdi and dialplan to simple get the analog phone to work. Can someone point me to a example of what I am trying to accomplish? Not wanting handholding but a push in the right direction. 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 -- _ -- 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] ISDN PRI configuration
Try this instead: http://www.ahk.com/t1_cable.html That cisco link does not specify the cable itself, but only the pin outs. True T1 cable has a foil shield around each pair, also called ABAM cable in the telco world. Ethernet cable is twisted pair without any shielding between pairs. And one shield around all the pairs is not the same as ABAM. Lyle Giese LCR Computer Services, Inc. On 12/08/11 10:53, Carlos Alvarez wrote: A T1 cable according to this spec: http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml Crossing the 1/2 to 4/5 if needed. On Thu, Dec 8, 2011 at 9:37 AM, Olivier oza_4...@yahoo.fr mailto:oza_4...@yahoo.fr wrote: 2011/12/8, Carlos Alvarez car...@televolve.com mailto:car...@televolve.com: I am not Kevin, but I'll tell you that I will not EVER use an Ethernet cable for T1 again. Kevin and I have discussed this at length, and the should work plays out poorly in the real world, or at least mine. I've had it be fine, and had major problems. I can't even find a pattern to it, like length of cable. In a colo cabinet that was direct-connected to a carrier, it worked great for years and then one day...no T1. Just gone. Go down there and put in a real T1 cable, came right up, still up years later. I usually make my own, which type of cable are you then using ? since they are so expensive to buy. I just connect the four needed pins, pretty easy to do if you're not trying to stuff all eight wires into the connector. On Thu, Dec 8, 2011 at 5:57 AM, Tony Mountifield t...@softins.co.uk mailto:t...@softins.co.uk wrote: In article 4ee0b0e2.3050...@digium.com mailto:4ee0b0e2.3050...@digium.com, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: As I said before... an Ethernet cable will work nearly all the time, and at a 5m length it's probably fine. Kevin, under what circumstances would an Ethernet cable potentially not work with T1/E1? And in those circumstances, what should be used instead? I'm wondering because I had never realised it was an issue until you said. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk mailto:t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org mailto:t...@mountifield.org - http://tony.mountifield.org -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 tel:602-889-3003 -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Log for voicemail to email?
On 09/20/11 17:53, Kevin Oravits wrote: I am having a problem with one of my sites where they are not receiving the voicemail to email. I’ve done a lot of troubleshooting and can’t find the issue. It would be helpful if there was a log I could look at so that I could see perhaps where the email is being rejected. Does anyone know of a log that runs on Asterisk that would have this history? I’m running Asterisk 1.6 on CentOS 5.6. The server is not behind a firewall, the Firewall on the box is disabled, SELinux is disabled and I’ve added the IP to our filters. Oddly, we have the same setup at other sites but this is the only site it is not working at. Any ideas would be great. Thanks, *Kevin * -- /var/log/mail on any of the SuSE or RedHat boxes I have looked at. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] Need a volunteer for a Patch
On 08/03/11 09:49, Venefax wrote: I tried te route of using iptables and at top production time, it eats 5% of my server, brining it to 95+ CPU usage. Clearly, not an option. I need a patch for chan_sip that when alwaysauthreject=yes does not respond to any REGISTER packet if the username does not exists. I hope that Digium would include this otr similar option in the source code. Alternatively, a new option can be created in sip.conf. I am offering no money for this patch. I think all the community needs this to survive the attack of the evil men from shadowlands. Another nice patch that I already wrote partially, is for cdr_addons_mysql, but it should be included in all cdr-collecting technologies. I just do not save to the database any call that is not connected. This is NOT the same as setting the option at the cdr.conf level. Each cdr technology needs this option as well. I need to save all calls to my cdr_odbc, for ASR calculations, but it is useless to store un-connected calls to mysql, because I use it only as a backup cdr, in case my external SQL Server blows up or has a problem, which happens often. What I did was to hard code this option in the source code, but not including any checkin for a cdr_sql.conf, since I am not a C programmer. With your option turned on, evil ones will again be able to enumerate valid usernames. To keep them guessing, you give them the same answer if the user name does not exist or if they gave you a bad password. But with your option turned on, they will know if they have a valid user name or not. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] use dahdi for local terminal modem access?
On 07/22/11 22:47, William Stillwell wrote: Um, no VOIP involved here. Wrong. What do you think Asterisk is? Chopped meat? It's a VoIP switch. All traffic inside Asterisk is VoIP. I have an asterisk server with 2 23B+D PRI's I want to telnet/ssh into the asterisk server, and make an outbound call serial based modem/terminal connection (Like the 80/90's BBS Days). No TCP/IP or PPP or crazyness (ie, dialing into a Modem set to AA hooked to a Cisco Console Port) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Friday, July 22, 2011 8:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] use dahdi for local terminal modem access? On 07/22/11 18:13, William Stillwell wrote: I have some terminals that have phone lines. One of my tech had an idea of using IAXmodem or something similar to use existing PRI/DAHDI Trucks for dial out via the asterisk/Linux console. Anybody ever heard of doing this? I would think maybe would use iaxmodem maybe and a shell terminal app? (basically I'm dialing into a remote access device that uses a pots like for remote administration, and don't want to string a channel bank off my asterisk box, and a hook to a modem) -- Depends on your expectation. Because of compression in the codecs, it will be hard to get fast dialup. If you mean ssh or telnet, it might work. If you mean vnc or RDP over this, you may not get enough usable bandwidth to do that. Given this, I have in an emergency dialed into a RAS server via a VoIP line. My laptop connected at 14,400bps. All I needed to do was telnet into an APC masterswitch to toggle power on one outlet. It worked. I was surprised at getting a 14,400bps connect. I was not expecting that high and really did not need that high. 300 baud probably would have been fast enough to telnet into an APC masterswitch. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] use dahdi for local terminal modem access?
On 07/22/11 18:13, William Stillwell wrote: I have some terminals that have phone lines. One of my tech had an idea of using IAXmodem or something similar to use existing PRI/DAHDI Trucks for dial out via the asterisk/Linux console. Anybody ever heard of doing this? I would think maybe would use iaxmodem maybe and a shell terminal app? (basically I’m dialing into a remote access device that uses a pots like for remote administration, and don’t want to string a channel bank off my asterisk box, and a hook to a modem) -- Depends on your expectation. Because of compression in the codecs, it will be hard to get fast dialup. If you mean ssh or telnet, it might work. If you mean vnc or RDP over this, you may not get enough usable bandwidth to do that. Given this, I have in an emergency dialed into a RAS server via a VoIP line. My laptop connected at 14,400bps. All I needed to do was telnet into an APC masterswitch to toggle power on one outlet. It worked. I was surprised at getting a 14,400bps connect. I was not expecting that high and really did not need that high. 300 baud probably would have been fast enough to telnet into an APC masterswitch. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] Re : Re : Re : Direct RTP with Asterisk
The only way this will work is to remove NAT from this scenerio. And it's not Asterisk's fault per se. The phones are built 'that way' also. That's why other free providers don't use SIP phones, but build their own client software. The others are trying to tell you SIP/RTP doesn't work the way you want it to. Lyle Giese LCR Computer Services, Inc. On 06/20/11 10:05, Sagbo Romaric wrote: Ok, thanks, Can you help me to have this kind of rules ? I try with iptables without success. Best, Romaric SAGBO *De :* Paul Hayes p...@provu.co.uk *À :* asterisk-users@lists.digium.com *Envoyé le :* Lun 20 juin 2011, 16h 39min 32s *Objet :* Re: [asterisk-users] Re : Re : Direct RTP with Asterisk On 20/06/11 13:18, Eric Wieling wrote: If you can't ping between the two end points, then you can't do direct RTP. precisely. If 10.10.9.1 isn't reachable from the network that 10.10.8.1 is on then 10.10.8.1 isn't going to be able to send RTP to 10.10.9.1. You need to add routes to the routers on both networks telling them how to reach the other networks. cheers, Paul -- _ -- 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 + VOSP account working configuration?
Gilles wrote: On Tue, 14 Dec 2010 16:56:14 +0100, Gilles codecompl...@free.fr wrote: PS: Here's what I'm thinking of using: At this point, Asterisk seems to register OK with my VOSP, but when I call the number from my cellphone, I get this error: NOTICE[88]: chan_sip.c:14033 handle_request_invite: Call from 'myvospaccount' to extension 's' rejected because extension not found. Incidently, how does Asterisk know how to link calls from the VOSP to an extension in the dialplan? Here's what I'm using: ; sip.conf [general] port = 5060 bindaddr = 0.0.0.0 ;deny=0.0.0.0/0 ;permit=IP address of VOSP server externip=my public IP address localnet=192.168.0.0/24 nat=yes ;all RTP packets go through Asterisk canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm ;incoming calls from VOSP ;can't use s extension? context = vosp-incoming register = myvospaccount:mypas...@myvosp.com ; extension.conf [general] static=yes writeprotect=yes clearglobalvars=no autofallthrough=yes [vosp-incoming] exten = s,1,Dial(SIP/6011) exten = s,n,Hangup Thank you. You are setting up a SIP trunk from your VOSP provider(whatever VOSP is). It dials your phone number. So whatever you dial from your cell phone is the extension that this trunk should land at. 's' is not an extension. It's a placeholder for the steps in your dial plan. For instance if my phone number with my provider is 815 555 1212, then I need an extension 811212. I would use: [inbound] exten = 811212,1,answer exten = 8151212,2,Goto(mainmenu,s,1) exten = 811212,3,hangup Lyle Giese LCR Computer Services, Inc. -- _ -- 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] DAHDI phantom pickup when ringing
Jonathan Hunter wrote:
On 24 November 2010 01:20, Lyle Giese l...@lcrcomputer.net
mailto:l...@lcrcomputer.net wrote:
Post the revelent portions of your extension.conf. Maybe you have
a logic error somewhere.
Thanks Lyle.
My extensions.conf is fairly simple in this regard; I use macro-stdexten:
[macro-stdexten];
exten = s,1,NoOp('${CALLERID(NAME)}' [${CALLERID(NUM)}] calling
[${ARG1}])
exten = s,n,Set(MBOXCONTEXT=)
exten = s,n,Dial(${ARG1},30) ; Ring the interface,
30 seconds maximum
exten = s,n,MailboxExists(${macro_ext...@${mboxcontext})
exten = s,n,NoOp(Got mailbox status of '${VMBOXEXISTSSTATUS}')
exten =
s,n,GotoIf($[${VMBOXEXISTSSTATUS}=SUCCESS]?s-Voicemail,1:s-NOANSWER,1)
and it is called with
SIP/DAHDI/1r1DAHDI/3r1DAHDI/5r1DAHDI/7r3SIP/SIP/SIP/SIP/SIP/DAHDI/2DAHDI/4DAHDI/6
Have you tried to move the set from channel 5 to 8 and 7 to 9? (to
see if one or two of the fxs channels have gone bad in the chan bank?)
Good idea, thank you - I will try this tonight.
It could also be a power supply issue inside the Zhone that tries
to 'trip' the ringing.
Hmm - not sure how I might determine whether this is the case or not..
It only seems to occur on some channels, at the moment.
Thinking on this, if the power supply is going bad, reducing the number
of DAHDI channels in ringing state may help. I am an old telco guy
having spent 23 years working for the biggest telco in the US in their
CO's. I tend to think something funny with the channel bank or the
channel units. Seen that happen many times working for them.
I assume the wiring is good and not 'wet'. If it was underground or in
a damp environment...
I would go back to thinking chan bank with FXS channel units, not DAHDI
channels. It will focus the attention where you have power supplies(-24
or -48 volt talk battery and ringing current with trip battery
super-imposed) and all the electronic things that can go wrong with that
and the detecting of offhook state. It would be easy for the
electronics to think the phone was offhook when ringing, but not when
supplying only talk battery when the channel units or power supply goes
flakey.
Lyle Giese
Thanks,
Jonathan
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Re: [asterisk-users] DAHDI phantom pickup when ringing
Jonathan Hunter wrote: On 21 November 2010 23:13, Jonathan Hunter jmhunt...@gmail.com mailto:jmhunt...@gmail.com wrote: I've been experiencing trouble with my DAHDI channels for some time and have finally decided to try and resolve the issue. Essentially, the problem I am having is that when a call comes in, and my DAHDI phones therefore ring, Asterisk thinks that one of the handsets has picked up to answer the incoming call - whereas in actual fact it is still on hook. The call then gets instantly dropped (the phone is on-hook, after all), and the caller has to redial. There has been no reply on this for a few days - is there a more appropriate forum I should be utilising, or is it just that nobody else has had these issues? Thanks, Jonathan -- If we knew what it was we were doing, it would not be called research, would it? - Albert Einstein Post the revelent portions of your extension.conf. Maybe you have a logic error somewhere. Have you tried to move the set from channel 5 to 8 and 7 to 9? (to see if one or two of the fxs channels have gone bad in the chan bank?) It could also be a power supply issue inside the Zhone that tries to 'trip' the ringing. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] Is this a DDoS to reach Asterisk?
Bruce B wrote: Hi Everyone, I have pfSense running which supplies Asterisk with DHCP. I had some testing ports opened for a web server which I have totally closed now but when I chose option 10 (filter log) on pfSense I get all of this type of traffic (note that it was only 1 single IP and once I blocked that one it was like opening a can full of bees with all different IPs): tcpdump: WARNING: pflog0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on pflog0, link-type PFLOG (OpenBSD pflog file), capture size 96 bytes 00 rule 70/0(match): block in on vr1: 221.132.34.165.33556 69.90.78.53.52229: tcp 20 [bad hdr length 0 - too short, 20] 6. 239658 rule 70/0(match): block in on vr1: 121.207.254.227.6667 69.90.78.38.3072: tcp 24 [bad hdr length 0 - too short, 20] 7. 986724 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 867707 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 799337 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 931814 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 574556 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 956066 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 598334 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 20 [bad hdr length 8 - too short, 20] 072759 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 109451 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 731065 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 16 [bad hdr length 12 - too short, 20] 159413 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 374293 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 16 [bad hdr length 12 - too short, 20] 10. 234202 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 2. 985558 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 13. 236084 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 2. 982122 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 18. 493312 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 2. 477084 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 9. 92 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: tcp 16 [bad hdr length 12 - too short, 20] 1. 216002 rule 70/0(match): block in on vr1: 172.168.0.4.1568 69.90.78.49.445: [|tcp] 321600 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: tcp 20 [bad hdr length 8 - too short, 20] 1. 383839 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: [|tcp] 1. 466115 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: [|tcp] 7. 977140 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 2. 920013 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 29. 032839 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 2. 996906 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 62. 079279 rule 70/0(match): block in on vr1: 82.165.131.28.6005 69.90.78.47.1024: [|tcp] 34. 224871 rule 67/0(match): block in on vr1: 77.34.234.241.1899 69.90.78.43.445: [|tcp] 3. 006367 rule 67/0(match): block in on vr1: 77.34.234.241.1899 69.90.78.43.445: [|tcp] 20. 274886 rule 67/0(match): block in on vr1: 66.211.120.62.1132 69.90.78.55.445: [|tcp] 2. 893859 rule 67/0(match): block in on vr1: 66.211.120.62.1132 69.90.78.55.445: [|tcp] 28. 739620 rule 67/0(match): block in on vr1: 117.197.247.151.1042 69.90.78.55.445: [|tcp] 2. 936286 rule 67/0(match): block in on vr1: 117.197.247.151.1042 69.90.78.55.445: [|tcp] 1. 207250 rule 67/0(match): block in on vr1: 118.171.176.188.42965 69.90.78.43.445: [|tcp] 3. 015370 rule 67/0(match): block in on vr1: 118.171.176.188.42965 69.90.78.43.445: [|tcp] 7. 088359 rule 67/0(match): block in on vr1: 61.130.103.10 69.90.78.42 http://69.90.78.42: [|icmp] 11.
Re: [asterisk-users] Is this a DDoS to reach Asterisk?
Welcome to the Internet! It's a fact of life when having equipment connected to the Internet. The script kiddies are always probing and trying. Lyle Bruce B wrote: And that's the problem. There is no such service running or such port is not open. They only keep trying this for no reason. It might cost us bandwidth for no reason. In fact there is no open ports on our network whatsoever. Thanks On Mon, Nov 8, 2010 at 9:50 PM, Lyle Giese l...@lcrcomputer.net mailto:l...@lcrcomputer.net wrote: Bruce B wrote: Hi Everyone, I have pfSense running which supplies Asterisk with DHCP. I had some testing ports opened for a web server which I have totally closed now but when I chose option 10 (filter log) on pfSense I get all of this type of traffic (note that it was only 1 single IP and once I blocked that one it was like opening a can full of bees with all different IPs): tcpdump: WARNING: pflog0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on pflog0, link-type PFLOG (OpenBSD pflog file), capture size 96 bytes 00 rule 70/0(match): block in on vr1: 221.132.34.165.33556 69.90.78.53.52229: tcp 20 [bad hdr length 0 - too short, 20] 6. 239658 rule 70/0(match): block in on vr1: 121.207.254.227.6667 69.90.78.38.3072: tcp 24 [bad hdr length 0 - too short, 20] 7. 986724 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 867707 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 799337 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 931814 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 574556 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 956066 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 598334 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 20 [bad hdr length 8 - too short, 20] 072759 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 109451 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 731065 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 16 [bad hdr length 12 - too short, 20] 159413 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 374293 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 16 [bad hdr length 12 - too short, 20] 10. 234202 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 2. 985558 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 13. 236084 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 2. 982122 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 18. 493312 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 2. 477084 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 9. 92 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: tcp 16 [bad hdr length 12 - too short, 20] 1. 216002 rule 70/0(match): block in on vr1: 172.168.0.4.1568 69.90.78.49.445: [|tcp] 321600 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: tcp 20 [bad hdr length 8 - too short, 20] 1. 383839 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: [|tcp] 1. 466115 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: [|tcp] 7. 977140 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 2. 920013 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 29. 032839 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 2. 996906 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 62. 079279 rule 70/0(match): block in on vr1: 82.165.131.28.6005 69.90.78.47.1024: [|tcp] 34. 224871 rule 67/0(match): block in on vr1
Re: [asterisk-users] Asterisk to switch on electric heaters remotely?
Gilles wrote: Hello I'm sure someone has already tried this: I use a couple of electric heaters to heat my office. I'd like to somehow connect them to Asterisk so that I could switch them on remotely by either calling the IVR or sending an e-mail to the Asterisk host, so that the room is warm when I get to the office :-) Any information appreciated. Thank you. I use a linux box to control the hvac in my home using a QK108 instead of a conventional thermostat(on a forced air nat gas furnace). I use 1wire probes from www.hobbyboards.com to monitor temperature and humidity. I wrote custom perl cgi scripts to control this via a webpage. Up to you from there on how fancy you want to get. I suspose you could use an IVR system to reset the temperature... Lyle -- _ -- 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] How to test BRI lines energy saving mode ?
Olivier wrote: Hello, If my understanding is correct, these days it seems that many ISDN BRI lines are configured in energy saving mode in which signalling D-channel is dropped until a new call comes in. Is it possible to replicate this behaviour with Asterisk (when Asterisk is in NT mode and is seen as a public ISDN by another PBX, for instance) ? If not, would you it would be a useful addition to Asterisk ? Regards Energy saving??? I don't think so. If the D channel is down, how would I make an outgoing phone call? Something in this mode or your explanation just does not sound right... Lyle Giese LCR Computer Services, Inc. -- _ -- 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] rtp problem with 1.8.0-rdc1
Benny Amorsen wrote: cov...@ccs.covici.com writes: Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Why is it a problem? It sounds like Asterisk does silence suppression. /Benny 1) With no rtp traffic, the nat device will drop the connection in it's nat table and thus disconnecting the softphone from Asterisk. (after the router's timeout period of course) 2) The other issue is you are connected to a conference call and you want to mute your transmitter while listening to the conference. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] sip probe syntax
Matt Kershnar wrote: If anyone has any info on this it'd be much appreciated - haven't found much about this topic anywhere. We are setting up sip probe monitor to make sure that our Asterisk boxes are up and functional (or at least responding to the sip protocol) and we need to determine the appropriate probe syntax for the probe requests to the Asterisk boxes. These boxes are running on various platforms and asterisk versions so we'd like to keep this as universal as possible. If we need to be platform/version specific any advice would be helpful. Thanks! http://exchange.nagios.org/directory/Plugins/Network-Protocols/*-VoIP/SIP/check_sip-sipsak/details Lyle Giese LCR Computer Services, Inc. -- _ -- 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] redirect based on incoming number
Barry Fawthrop wrote:
How does one redirect calls based on incoming number or caller ID or the
lack thereof?
current I have for number 123-4567 that it redirects all 800 , 877 and
866 numbers to Voicemail directly.
If the primary area code is 352 then accept this and pass it to
extension
exten = 1234567/_352XXX,4,Dial(SIP/,240)
exten = 1234567/_800XXX,4,Voicemail(5...@default,b)
exten = 1234567/_866XXX,4,Voicemail(5...@default,b)
exten = 1234567/_877XXX,4,Voicemail(5...@default,b)
exten = 1234567/1800XXX,4,Voicemail(5...@default,b)
exten = 1234567/1866XXX,4,Voicemail(5...@default,b)
exten = 1234567/1877XXX,4,Voicemail(5...@default,b)
exten = 1234567/+1800XXX,4,Voicemail(5...@default,b)
exten = 1234567/+1866XXX,4,Voicemail(5...@default,b)
exten = 1234567/+1877XXX,4,Voicemail(5...@default,b)
exten = 1234567/_*1866876.,4,Voicemail(5...@default,b)
exten = 1234567/_+18668762996,4,Voicemail(5...@default,b)
Any help will be greatly appriecated
Thanks
[menu]
exten = s,n,Set(NPA=${CALLERID(num):0:3}); grab area code from caller id
exten = s,n,GotoIF($[ ${NPA} = 800 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 888 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 877 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 866 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 855 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 844 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 833 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 822 ]?marketeer)
exten = s,n(marketeer),Set(TIMEOUT(digit)=6); allow humans to bypass
drop into VM
exten = s,n,Set(TIMEOUT(response)=10);
exten = s,n,Set(CALLERID(num)=51${CALLERID(num)})
exten = s,n,Background(missingcallerid); Dial 111 to actually ring a phone
exten = s,n,Voicemail(u111); no digits dialed drop into VM
exten = s,n,Hangup
Lyle
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Re: [asterisk-users] Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1
bruce bruce wrote: I am not sure why it would be sleeping. I have never dealt with putting a linux server to sleep. It is connected to a UPS, but I don't think it has been put to sleep by the UPS as the USB cable from UPS is not connected to it. Can you please elaborate on what you mean by AMI:Ping? Is there a service that you recommand that does this or are there any opensource monitoring tools out there that I can use? But my main question remains why there are no activities on 24th and 25th? This is what I see in the /var/log/messages.1: Jul 23 17:11:55 elastix last message repeated 20 times Jul 23 17:22:51 elastix last message repeated 38 times Jul 23 17:30:39 elastix last message repeated 26 times Jul 23 17:30:39 elastix last message repeated 45 times Jul 23 19:09:42 elastix ntpd[3113]: synchronized to 216.216.216.216, stratum 2 Jul 23 20:17:44 elastix ntpd[3113]: synchronized to 216.216.216.216, stratum 2 Jul 23 21:29:16 elastix dhclient: DHCPREQUEST on eth0 to 192.168.1.254 port 67 Jul 23 21:29:16 elastix dhclient: DHCPACK from 192.168.1.254 Jul 23 21:29:16 elastix dhclient: bound to 192.168.1.100 -- renewal in 37640 seconds. Jul 26 09:22:37 elastix syslogd 1.4.1: restart. Jul 26 09:22:37 elastix kernel: klogd 1.4.1, log source = /proc/kmsg started. Jul 26 09:22:37 elastix kernel: Linux version 2.6.18-164.el5 (mockbu...@builder16.centos.org mailto:mockbu...@builder16.centos.org) (gcc version 4.1.2 20080704 (Red Hat 4.1.2-46)) #1 SMP Thu Se$ Jul 26 09:22:37 elastix kernel: BIOS-provided physical RAM map: Morning of the 26th at 9:22 the server was restarted because it was un-reachable from outside and hence the restart log but where is the 24th, and 25th? Thanks, Bruce On Thu, Jul 29, 2010 at 9:10 AM, Paul Belanger paul.belan...@polybeacon.com mailto:paul.belan...@polybeacon.com wrote: On Wed, Jul 28, 2010 at 9:06 PM, bruce bruce bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: See the jump from Jul 23rd to Jul 26th. Is this an indication of Asterisk being down? No, it just means there was no logger activity for those days. You need to add a monitoring solution to your Asterisk box (IE: AMI: Ping). -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com mailto:paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com http://blog.polybeacon.com -- It's 'well known' that Asterisk gets confused and runs around in a very tight loop when DNS resolution is failing. Asterisk does a lot of DNS queries and when the Internet goes down, that puts Asterisk into a loop. Depending on your machine, I am guessing that Asterisk locked up or dropped out on the 23rd and the restart on the 26th brought it back to life. Nagios is a good choice for monitoring servers and services. I use it here to monitor all the servers and SIP on my Asterisk box. Lyle Giese LCR Computer Services, Inc. -- _ -- 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 stopped after Internet connection dropped ?! Asterisk 1.4.26.1
Lyle Giese wrote: bruce bruce wrote: I am not sure why it would be sleeping. I have never dealt with putting a linux server to sleep. It is connected to a UPS, but I don't think it has been put to sleep by the UPS as the USB cable from UPS is not connected to it. Can you please elaborate on what you mean by AMI:Ping? Is there a service that you recommand that does this or are there any opensource monitoring tools out there that I can use? But my main question remains why there are no activities on 24th and 25th? This is what I see in the /var/log/messages.1: Jul 23 17:11:55 elastix last message repeated 20 times Jul 23 17:22:51 elastix last message repeated 38 times Jul 23 17:30:39 elastix last message repeated 26 times Jul 23 17:30:39 elastix last message repeated 45 times Jul 23 19:09:42 elastix ntpd[3113]: synchronized to 216.216.216.216, stratum 2 Jul 23 20:17:44 elastix ntpd[3113]: synchronized to 216.216.216.216, stratum 2 Jul 23 21:29:16 elastix dhclient: DHCPREQUEST on eth0 to 192.168.1.254 port 67 Jul 23 21:29:16 elastix dhclient: DHCPACK from 192.168.1.254 Jul 23 21:29:16 elastix dhclient: bound to 192.168.1.100 -- renewal in 37640 seconds. Jul 26 09:22:37 elastix syslogd 1.4.1: restart. Jul 26 09:22:37 elastix kernel: klogd 1.4.1, log source = /proc/kmsg started. Jul 26 09:22:37 elastix kernel: Linux version 2.6.18-164.el5 (mockbu...@builder16.centos.org mailto:mockbu...@builder16.centos.org) (gcc version 4.1.2 20080704 (Red Hat 4.1.2-46)) #1 SMP Thu Se$ Jul 26 09:22:37 elastix kernel: BIOS-provided physical RAM map: Morning of the 26th at 9:22 the server was restarted because it was un-reachable from outside and hence the restart log but where is the 24th, and 25th? Thanks, Bruce On Thu, Jul 29, 2010 at 9:10 AM, Paul Belanger paul.belan...@polybeacon.com mailto:paul.belan...@polybeacon.com wrote: On Wed, Jul 28, 2010 at 9:06 PM, bruce bruce bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: See the jump from Jul 23rd to Jul 26th. Is this an indication of Asterisk being down? No, it just means there was no logger activity for those days. You need to add a monitoring solution to your Asterisk box (IE: AMI: Ping). -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com mailto:paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com http://blog.polybeacon.com -- It's 'well known' that Asterisk gets confused and runs around in a very tight loop when DNS resolution is failing. Asterisk does a lot of DNS queries and when the Internet goes down, that puts Asterisk into a loop. Depending on your machine, I am guessing that Asterisk locked up or dropped out on the 23rd and the restart on the 26th brought it back to life. Nagios is a good choice for monitoring servers and services. I use it here to monitor all the servers and SIP on my Asterisk box. Lyle Giese LCR Computer Services, Inc. While the above comment about DNS holds, I also realized that most likely your Asterisk machine lost it's only ip address when the DSL went down. That may also have caused Asterisk to exit. I think most(if not all) admins here would never have a dynamic ip address on an Asterisk server. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] Question regarding SMS(), SMSQ, SMSC
Maybe you need to read the man page for qpage. The qpage client can send the page to an SNPP server over TCP/IP. Lyle AMARDEEP SINGH wrote: Our SMS-gateway is not PSTN accessible. On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese l...@lcrcomputer.net mailto:l...@lcrcomputer.net wrote: AMARDEEP SINGH wrote: Hello All, Scenario: -We use asterisk as voicemail server for our cellular network. Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. -Extensions in * are virtual, just for leaving and accessing voicemail. Requirement: Asterisk to send SMS to cell switch(running SMSC) on reception of new voicemail. Pointers required from Maillist users: -How can I make * talk to SMSC(ip address:port). -Anyone using similar topology? -there are not enough examples/man/maillist of using app_sms(), smsq. Thanks: -A qpage? -- _ -- 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] Question regarding SMS(), SMSQ, SMSC
qpage -s snppserver.example.com -p lyle -f lyle test page AMARDEEP SINGH wrote: Do you have working script? On Fri, Jul 23, 2010 at 10:14 AM, Lyle Giese l...@lcrcomputer.net mailto:l...@lcrcomputer.net wrote: Maybe you need to read the man page for qpage. The qpage client can send the page to an SNPP server over TCP/IP. Lyle AMARDEEP SINGH wrote: Our SMS-gateway is not PSTN accessible. On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese l...@lcrcomputer.net mailto:l...@lcrcomputer.net mailto:l...@lcrcomputer.net mailto:l...@lcrcomputer.net wrote: AMARDEEP SINGH wrote: Hello All, Scenario: -We use asterisk as voicemail server for our cellular network. Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. -Extensions in * are virtual, just for leaving and accessing voicemail. Requirement: Asterisk to send SMS to cell switch(running SMSC) on reception of new voicemail. Pointers required from Maillist users: -How can I make * talk to SMSC(ip address:port). -Anyone using similar topology? -there are not enough examples/man/maillist of using app_sms(), smsq. Thanks: -A qpage? -- _ -- 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] Question regarding SMS(), SMSQ, SMSC
AMARDEEP SINGH wrote: Hello All, Scenario: -We use asterisk as voicemail server for our cellular network. Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. -Extensions in * are virtual, just for leaving and accessing voicemail. Requirement: Asterisk to send SMS to cell switch(running SMSC) on reception of new voicemail. Pointers required from Maillist users: -How can I make * talk to SMSC(ip address:port). -Anyone using similar topology? -there are not enough examples/man/maillist of using app_sms(), smsq. Thanks: -A qpage? -- _ -- 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] Still sipping frustration - only getting state ACK
Julien Claassen wrote: Hello everyone! I still am not much further along with my sip calling. I changed my sip.conf taking suggestions from the net (voip-info.org in particular). I changed iptel's position from friend to peer. I turned on and off nat, I chose different codecs in first place, entered my outward IP as fromdomain and uncommented the register directive with correct values. All I get is two registrations now, but no calls. get a registration effort every 225secs and it succeeds. But when I make a call; channel originate sip/iptel-out/e...@iptel.org Application playback vm/net_ring The call is onlyleft in state ACK for a while. Then asterisk tells me, that it is destroying the sip dialog (long ID) INVITE. Question: Might it be a problem, that my system only knows itself as 192.168.*. Do I need to set something else than externip? Does the server see your sip client at 192.168.*.*? that would be a problem. Might it be, that my router really blocks certain ports? I can't check it, since it's heavily javascript based and, since I'm blind and the accessibility software for the GUI never really worked on this distro, I don't have a browser to look at it. It's possible that the router is not SIP friendly or there is a setting to allow sip on it. I can not tell as I don't know what router you are using. Do I need to forward port 5060 to my machine specifically (like it is needed for SSH's port 22), or is the mechanism based on: I talk first and the sever gets back to me based on that. Should not need any forwards. However the router could be firewalling some ports, like the rtp ports. You need to ask what ports are needed for rtp. Lyle Giese LCR Computer Services, Inc. This configuration worked for googletalk. I admit, there were problems, but calls were coming through from both sides. Please can someone help me clear up this mess. I'm completely frustrated and don't know what else to do, where else to look. Kindly yours and thanks in advance JUlien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] SpiderMux?
Tim Nelson wrote: Greetings all- I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks rather interesting. Has anyone used one? Where did you purchase it? Pricing? Operational issues? http://spidermux.com/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 A couple of things bother me about their webpage. The link for the manufacturers home page goes to an expired domain name. And the price list page is dated in 2006. -- _ -- 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] Changing storm-prevention behaviour in logger.conf
Tilghman Lesher wrote: On Saturday 17 April 2010 16:14:23 Remco Bressers wrote: Dear List, According to https://issues.asterisk.org/view.php?id=14905 there is a storm prevention mechanism in newer Asterisks. If i look in my logfile, i see : [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Registration from ' sip:x...@xxx.xxx.xxx.xxx' failed for 'xx.xx.xx.xx' - Wrong password [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message repeated 3 times This IS a good thing to do, but i want to disable this behaviour. We are using fail2ban to ban scripts and people from the Asterisk system. On version 1.4.23 this worked fine, but now this mechanism is in place, i cannot use fail2ban anymore. Is there any option to disable this behaviour, or even better, add it to logger.conf so anybody can decide what to do? I just want all logging and it seems impossible now. Maybe a patch on the source? That's not Asterisk doing that. That's your system logger. AFAIK, there's no way to turn that off, as it's a defense mechanism against an attacker filling your disks, causing lost messages and possible crashes (on some platforms). If running syslog-ng, check syslog-ng.conf and the summary option. Setting summary to 0 turns off that behavior. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] indications.conf
Patience is a virtue. Demanding answers or responses is a sure fire way to get ignored, esp since you waited only a few hours for a response. Here's it's Sunday. Traffic levels are down over the weekend as most list users here are doing family things instead of their jobs. Besides, this list is a free resource where the populace participates because they want to, not because they are paid to answer questions. I did not answer because I am not running 1.6.x yet and I have no need for tones other than US. Lyle Ciprian ARSENIE wrote: Nobody?? The problem is only in asterisk 1.6.2.X in asterisk 1.6.0.x is working ??? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cipr...@carsenie.ro Sent: Sunday, March 07, 2010 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] indications.conf hi I have problelm with an asterisk 1.6.2.5 tarbal compiled on CentOS 5.4 and try to change tones in indications.conf but any setting i have made has no effect. the tones are by default U.S. and i need to change to hungarian or greece -- _ -- 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] Security Logging
Warren Selby wrote: On Tue, Feb 9, 2010 at 5:54 PM, Lyle Giese l...@lcrcomputer.net mailto:l...@lcrcomputer.net wrote: Here's a start for you, just run from cron once a day: Lyle So basically, nothing built into asterisk that already provides security logging mechanisms? Maybe I'm using the wrong term; In Windows, I think it would be called Security Auditing, successful / unsuccessful login attempts that get recorded in the Windows Event Viewer in the security log. These login attempts (whether successful or not) are recorded, and you get the IP address of the workstation attempting the login, the username used, and whether or not it was successful. A log dedicated just to security auditing (or a new option in /etc/logger.conf that adds this functionality (say, messages = notice,warning,error,verbose,security) seems like it would be a nice addition to asterisk. I've already got tools that can monitor log files and create bans based on failed login attempts...but I don't always seem to see login failures in the asterisk messages log. I recall from Astricon 2009, Russel and Kevin (I think) commenting on security features in asterisk and not sure how much to include (i.e automatically banning people based on failed login attempts being a process asterisk controls or just simply logs so that another tool can do the banning, etc). I just don't remember if there was any followup to those discussions. -- Thanks, --Warren Selby http://www.selbytech.com I think that is the problem. Nobody can agree on how it should be implemented. So just log the events and the user/admin find and use a log analyzer or build your own tools for those that want/need such. Lyle -- _ -- 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] Security Logging
Warren Selby wrote:
Hello list,
I've got a client who's weak sip passwords are being guessed by remote
entities who then connect to their server and use it to wardial large
swaths of numbers. When they start receiving complaints, they call me
and I add the ip address of the remote user to the iptables drop list.
At the same time, my own personal asterisk server, using strong sip
passwords, has seen connections from remote entities. I'm not sure
how these passwords were guessed (or even if they were guessed), as
they were at a minimum 10 characters long, not based on dictionary
words, and used numbers, letters, and symbols.
Is there some logging capability that allows me to see every IP
address of every sip registration attempt, along with details about
the sip reg attempt (I.e user name tried, success or failure, user
agent, etc). I haven't found a way to do this yet, I'm hoping I've
just missed something simple?
Thanks,
Warren Selby
Here's a start for you, just run from cron once a day:
Lyle
#!/usr/bin/perl
$mess_log = /home/asterisk/log/asterisk/messages;
$event_log = /home/asterisk/log/asterisk/event_log;
$queue_log = /home/asterisk/log/asterisk/queue_log;
$cdr_log = /home/asterisk/log/asterisk/cdr-csv/Master.csv;
$vm_dir = /home/asterisk/spool/asterisk/voicemail/default/;
$sendmail = /usr/sbin/sendmail -t ;
$ast_log = /home/asterisk/log/asterisk/messages;
open astlog, $ast_log || die Could not open Asterisk logs\n;
open ast_mail, | $sendmail;
print ast_mail To: email1\n;
print ast_mail From: root\n;
print ast_mail Subject: Asterisk passwd fail log\n;
open ast_mail2, | $sendmail;
print ast_mail2 To: email1\n;
print ast_mail2 From: root\n;
print ast_mail2 Subject: Asterisk bad SIP number log\n;
while (astlog) {
chomp;
$ln = $_;
if (index($ln,password) ne -1) {
print ast_mail $ln . \n;
}
if (index($ln,matching) ne -1) {
print ast_mail2 $ln . \n;
}
}
close astlog;
close ast_mail;
close ast_mail2;
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Re: [asterisk-users] sip to dahdi and billsec
Uros Djokic wrote: It entirely depends on the technology used to interface to the PSTN. You have not specified what technology/hardware you are using to connect to the PSTN. For instance if you are using POTS(plain old telephone service - analog copper fed lines), you do not get answer supervision back from the telco.-- I am using tdm400 card with one fxo port. I am using analog line so I guess it's POTS analog cooper fed line. So it is impossible to distinguish ringing from talking and billsec must start when ringing begin due missing answer supervision from telco ? Thanks for reply Answer supervision is missing in one sense. It was never part of the spec for this type of telco line. You will have to forgive calls under x number of seconds in duration as if they never occured or get a different type of connection from your telco that will include answer supervision. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] sip to dahdi and billsec
Uros Djokic wrote:
Hi,
My costumers are logged in on my Asterisk PBX through XLite Softphone
(SIP). My server is
connected to PSTN. Problem is when SIP phone calls ordinary phone via
dahdi I get
DAHDI/1-1 ANSWERED SIP/number-number and billsec field from cdr is
start counting.
Is it normal behavior ? Can I change that ?
So channel gets in ANSWERED state and billsec starts as soon as line
starts
to ring even if no one really pick up ordinary phone and costumer did
not talk to anyone.
That leads to problem that costumers will be billed even if they did
not make a real
conversation.
How can I avoid that behavior and set asterisk to start counting
billsecs after
someone really pick up the phone on the other side ?
How can I distinguish real (talking to) call from just ring (no real
answer call)
when both are in state ANSWERED ?
I tried with timeout 20 in Dial command but since channel is
answered when it
starts to ring timeout is not doing what I want.
Here is my Dial command:
exten = _X.,n,Dial(dahdi/g0/${EXTEN},20,L(${Limit}:6:2)hH)
It works very good in case ordinary phone calls sip (for incoming
calls from PSTN)
because I need to click answer on xlite to move call in state ANSWERED
so if I don't
click it is not answered and timeout works.
Can you help me with that ?
Thanks,
Uros
--
Use Free Software http://www.fsf.org/
---
Four essential software freedoms:
1) To study source code
2) To copy program
3) To modify source code
4) To redistribute modified program under condition that new user has
all 4 freedoms.
Richard M. Stallman
It entirely depends on the technology used to interface to the PSTN.
You have not specified what technology/hardware you are using to connect
to the PSTN.
For instance if you are using POTS(plain old telephone service - analog
copper fed lines), you do not get answer supervision back from the telco.
Lyle Giese
LCR Computer Services, Inc.
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Re: [asterisk-users] Virtual Asterisk Installation
Jeff LaCoursiere wrote: On Thu, 21 Jan 2010, Gergo Csibra wrote: Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote: Forget about virtualization! ... Virtualisation is nice for test-setups, but thats it. for any real job it's a major pain in the ass and makes stuff bork beyond imagination. Well. Why do you use computer? There're slide-rule. You can calculate anything with that... Pretty crappy analogy. Just because you *can* do something doesn't mean it is production ready. But then the OP said it wasn't all that important, so I would say go Xen and tell us how it works out. I think you will only have trouble with conferencing, and maybe not even then if the machine is beefy enough and unloaded. Monitoring servers are usually pretty unloaded. I'm playing a lot with OpenVZ, but you won't have access to your PSTN hardware... at least I haven't been able to make that part work. j Asterisk and monitoring are time sensitive applications. VM's are not good canidates for these types of services. Go to the MRTG discussions and you will get the same answer, stay away from VM. The time shift that VM's introduce cause huge issues when mapping time sensitive data. And Asterisk is time sensitive. A webserver or database server are not time sensitive applications where time shifts of a few milliseconds are not noticed. But with Asterisk if the time is shifting 20 or 30 ms frequently, it will cause all sorts of issues. Use VM's where and when useful. This scenerio is not a good candiate for virtualization. Lyle -- _ -- 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] Beginners Guide to setting up a Call Centre
Peter Childs wrote: Ok this has Probably been put to bed several time but never mind. Elastix, Trixbox, or AsterixNow, or DIY (ie Ubuntu or whatever installed with OpenPBX, Asterix etc by hand) I've got a new server to run Asterix on and want to get it working quickly and yet be configurable in the future with out having to reisntall and start again regally. Currently no VoIP hardware but that will come once I prove the concept. I guess Oh the machine does not have a CD Rom Drive so a network/USB install would be nice.. But I guess I can open the case and plug one in for installation if I must! (Says he who has just installed Ubuntu over the network to check the computer works!) Peter. In regards to a CD/DVD drive, I have a small ISP farm for web/email hosting. I stopped putting cd/dvd readers in the servers about 2 years ago. All the new motherboards out there support booting from a USB drive, so why bother? Get one good DVD drive and put it in a case with a USB adapter in it and just plug it in when you need it. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] Changing ring cadence on FXS lines
Noah I. Engelberth wrote: Is there a way I can change the ring cadence on FXS lines on a system using a Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the customer has a few POTS phones in areas where they don't have data network services, so we're using the FXS lines to provide dialtone at those outbuildings. The old phone system would ring those phones with a short ring-short ring-pause cadence, which sounds louder to the users than Asterisk's default long ring-pause cadence. I tried setting a cadence line in chan_dahdi.conf and restarting Asterisk, and typing dahdi show cadences in the CLI after the restart showed my custom cadence, but the phones were still ringing long ring-pause. Can someone point me in the direction of what I'm doing wrong? http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels Lyle Giese LCR Computer Services, Inc. -- _ -- 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] Grandstream GXW-4004
hin lee wrote: I am consider replacing my TDM card for a FXS gateway. Anyone has any issues with the Grandstream GXW-4004 on Asterisk? I would like some feedback before I spend the $$ this device. http://www.voip-info.org/wiki/view/Grandstream+GXW-4004 Thanks! Just to be clear, the Grandstream gateway is used to interface analog telephones to Asterisk, not for bringing in outside dialtone from your local telco to Asterisk. Why not buy SIP phones instead? I have not used it, so I have no opinion on it, but whose TDM card are you using now. Lyle Giese LCR Computer Services, Inc. ___ -- 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] Can't restart asterisk from script
Doug Lytle wrote: Warren Selby wrote: On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca mailto:supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully I have the following cron job: /usr/sbin/asterisk -r -x 'restart when convenient' Doug You probably don't need the single or double quotes at all. I have never used any quoting in crontab. Lyle Giese LCR Computer Services, Inc. ___ -- 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] Failure of user registration with XLITE
/[r...@dhcppc0 asterisk]# vi extensions.conf [tutorial] exten = 1234,1,Dial(SIP,gianca)/ /exten = 12345,1,Dial(SIP,giusy) / Here the XLITE user data: /Display Name: gianca/ /Username: 1234/ /Password: pwd_gianca/ /Authorization User Name: 1234/ /Domain: 192.168.1.100/ http://192.168.1.100/ Your XLITE user name should be the same as the sip account name(gianca not 1234). And the extensions.conf should be: exten = 1234,1,Dial(SIP/gianca) giancarlo lombardo wrote: Ciao, the problem is still present, does anyone have some other suggestion ? Below the output of CLI with debug option on XLITE IP and show peers command: /dhcppc0*CLI --- SIP read from 192.168.1.116:14166 http://192.168.1.116:14166 --- REGISTER sip:192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14166;branch=z9hG4bK-d8754z-4d4ced5bca35b64c-1---d8754z-;rport Max-Forwards: 70 Contact: sip:1...@192.168.1.116:14166;rinstance=c18a16f442f17333 To: giancasip:1...@192.168.1.100 mailto:sip%3a1...@192.168.1.100 From: giancasip:1...@192.168.1.100 mailto:sip%3a1...@192.168.1.100;tag=be7e8a36 Call-ID: YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 0/ /- --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.116 : 14166 (NAT)/ /--- Transmitting (NAT) to 192.168.1.116:14166 http://192.168.1.116:14166 --- SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.1.116:14166;branch=z9hG4bK-d8754z-4d4ced5bca35b64c-1---d8754z-;received=192.168.1.116;rport=14166 From: giancasip:1...@192.168.1.100 mailto:sip%3a1...@192.168.1.100;tag=be7e8a36 To: giancasip:1...@192.168.1.100 mailto:sip%3a1...@192.168.1.100;tag=as0194534b Call-ID: YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE. CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0/ / Scheduling destruction of SIP dialog 'YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE.' in 32000 ms (Method: REGISTER) Really destroying SIP dialog 'YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE.' Method: REGISTER dhcppc0*CLI sip show peers Name/username HostDyn Nat ACL Port Status giusy/giusy(Unspecified)D 0 Unmonitored gianca/gianca (Unspecified)D 0 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] dhcppc0*CLI/ 2009/11/8 Ahmed Ossama ah...@master-zone.net mailto:ah...@master-zone.net Hello, Try this in X-Lite config section: /Display Name: gianca/ /Username: //gianca/ /Password: pwd_gianca/ /Authorization User Name: //gianca/ /Domain: 192.168.1.100 / Ahmed Ossama giancarlo lombardo wrote: Dear all, I'm setting up a connection via XLITE softphone and asterisk 1.4 but I get the error: /Registration error: 404 Not found/ Here my configuration file of asterisk: /[r...@dhcppc0 asterisk]# vi sip.conf [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial/ /[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial/ /[r...@dhcppc0 asterisk]# vi extensions.conf [tutorial] exten = 1234,1,Dial(SIP,gianca)/ /exten = 12345,1,Dial(SIP,giusy) / Here the XLITE user data: /Display Name: gianca/ /Username: 1234/ /Password: pwd_gianca/ /Authorization User Name: 1234/ /Domain: 192.168.1.100/ http://192.168.1.100/ Here the output of wireshark in between Xlite client and asterisk server: // /0040 2e 31 30 30 20 53 49 50 2f 32 2e 30 0d 0a 56 69 .100 SIP /2.0..Vi 0050 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 50 20 31 a: SIP/2 .0/UDP 1 0060 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a 35 34 30 92.168.1 .116:540 0070 35 30 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 62 50;branc h=z9hG4b 0080 4b 2d 64 38 37 35 34 7a 2d 32 34 32 38 38 65 37 K-d8754z -24288e7 0090 32 38 32 36 64 30 31 32 38 2d 31 2d 2d 2d 64 38 2826d012 8-1---d8 00a0 37 35 34 7a 2d 3b 72 70 6f 72 74 0d 0a 4d 61 78 754z-;rp ort..Max 00b0 2d 46 6f 72 77 61 72 64 73 3a 20 37 30 0d 0a 43 -Forward s: 70..C 00c0 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 31 32 33 ontact: sip:123 00d0 34 40 31 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a //4...@192.16/ mailto:4...@192.16 mailto:4...@192.16/ 8.1.116: 00e0 35 34 30 35 30 3b 72 69 6e 73 74 61 6e 63 65 3d 54050;ri nstance= 00f0 36 33 61 39 66 64 62 62 62 62 39
Re: [asterisk-users] outbound routing
Contexts. Put the 'Source channels' in different contexts. Lyle B.Masoud @ SH wrote: Can you tell me how on the first question? Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Sunday, November 08, 2009 10:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] outbound routing -- Sent from mobile device On Nov 8, 2009, at 2:13 PM, B.Masoud @ SH i...@saudihome.com wrote: I have 2 questions: 1. Can I make outbound route rule based on the Source Channel? Yes. 2. Can I auto change the outbound route based on time/Day of week? Yes. See GotoIfTime(). Any help very appreciated.. ___ -- 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: 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] interfacing asterisk with a legacy PBX
PATRICK KANGETHE wrote: I want to interface asterisk with a legacy pbx that has around 23 extensions through my 8 fxs card, how do i work around this? Hint: I have already terminated 8 extensions from the legacy PBX, i was thinking whether i can peer the extensions from the PBX i.e like 5 extensions be peered to one extension connecting to the fxs? How can i do this? Thanks in advance, Are you planning to get rid of the legacy PBX completely? Or is Asterisk going to be a second PBX? I am going to assume you are replacing the legacy PBX. You can setup analog extensions so that you have multiple phones on each FXS channel. But they will be like a party line. If you put 6 phones on one FXS, all 6 ring at the same time, only one person can use that extension at a time. However you can add SIP phones to Asterisk and each can have their own extension instead. It just requires cat 5 cable back to a switch for each phone. Lyle Giese LCR Computer Services, Inc. ___ -- 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] polarity on some channels
B.Masoud @ SH wrote: Hello, I have : answeronpolarityswitch=yes on chan_dahdi.conf but it's making all my lines answer on polarity reversal, this causes a problem for PSTN lines, so how can I set these lines to answer immediately (when it rings)? thanks ___ -- 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 Try turning off callerid. The 'standard' for POTS lines in the US is to put the caller id in between ring1 ring2. Asterisk waits for callerid before answering the line by default. usecallerid=off ___ -- 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] Sending Dialled number down a sip channel to a PBX
Ishfaq Malik wrote: Bumping this in the hope that it is seen by people who missed it before. Ishfaq Malik wrote: We have a customer who connects PBX boxes (Avaya etc.) to our asterisk server (1.4.17) as a SIP extension. This customer needs the dialled number sent to the PBX as well as number that the call is originating from so he can set up his own routing from his PBX box. I have tried setting both CALLERID(dnid) and CALLERID(rdnis) to the dialled number, though not at the same time but the customers PBX box does not pick up the dialled number setting. Has anyone got any experience in this? Thanks Ish I am no expert in this area, but my question would be 'Does sip support sending the called number on a trunk?'. Lyle ___ -- 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
Re: [asterisk-users] Asterisk on a Beagleboard?
Vincent wrote: Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you. ___ -- 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 128m of ram 256 m flash for the 'hard drive' is not much in either catagory. And ethernet is a USB addon, not on the board. ___ -- 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
Re: [asterisk-users] Inquiry:Problem with Call Parking
And now that the whole world of Asterisk has your sip user ids and passwords, you should change all of the passwords that are in that file and yes, change the passwords in all your phones. Lyle Giese LCR Computer Services, Inc. hadi motamedi wrote: Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney) john@compuware.com mailto:john@compuware.com wrote: Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? 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 hadi motamedi Sent: Tuesday, 1 September 2009 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inquiry:Problem with Call Parking Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- 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 ___ -- 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
Re: [asterisk-users] Problems sending voicemail emails
The receiving server does not ask for any user id or password. The protocal says, the sender has to just send the user or pass command with the data required. Try reading /var/log/mail(if you have access), at least that's where the outgoing mail logs on my servers are. Lyle Joan Antoni Terre wrote: Jonathan, now I've done telnet mx1.datagrama.net http://mx1.datagrama.net 25 And I've got: Trying 212.9.65.110... Connected to mx1.datagrama.net http://mx1.datagrama.net (212.9.65.110). Escape character is '^]'. 220 mailhub03.datagrama.net http://mailhub03.datagrama.net ESMTP Datagrama It looks as it has connected but has not asked for any user / Password mx1.datagrama.net http://mx1.datagrama.net is my ISP ESMT server. 2009/8/24 Jonathan Moore supermegat...@gmail.com mailto:supermegat...@gmail.com On Mon, Aug 24, 2009 at 9:25 AM, Joan Antoni Terrenebh...@gmail.com mailto:nebh...@gmail.com wrote: Hi Michelle, If I try telnet mx1.datagrama.net http://mx1.datagrama.net/ I have no answer, I get: Trying 212.9.65.110... ¿? telnet mx1.datagrama.net http://mx1.datagrama.net/ 25 that's a space, then the port, in this case, 25. is ms1.datagrama.net http://ms1.datagrama.net/ what you really want though? It looks like you're using mydomain.com http://mydomain.com/ as the domain in your asterisk configuration. Do you really intend to use mydomain.com http://mydomain.com/ ? -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- 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 ___ -- 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
Re: [asterisk-users] Looking for wisdom - One Asterisk system - Multi-incoming trunks
Steve Edwards wrote:
On Wed, 29 Jul 2009, Myles Wakeham wrote:
I have setup an Asterisk system for my home home office.
[snip]
The cost of all these lines with analog carriers was getting ridiculous,
so I'm moving over to a SIP carrier. I created one account for a single
phone number with a SIP carrier (BroadVoice)
[snip]
I've never used BroadVoice, so I have nothing good or bad to say about
them. I've used Vitelity.net for several years and am pleased with them.
I have a nominal monthly fee, pay per minute account. They get $1.49 a
month for a DID and $0.0144 per minute. You'd have to use about 2,600
minutes (about 44 hours) before it would cost as much as a $40 per month
analog. They have an unlimited inbound for $7.95 a month.
I started the process today to get our other phone numbers moved over to
BroadVoice.
[snip]
Vitelity.net charges $18 per number ported. I've never done this.
My approach is to have one trunk provided by the SIP provider. All
numbers are allocated to that trunk (BroadVoice let me do that when I
setup the number transfer). Asterisk receives an incoming call on that
trunk and determines the calling number that it was requesting (not sure
how to get this, but Broadvoice assured me I could). Anyway after
determining what the call was destined for, I then route the call to the
appropriate context in the extensions to handle it.
The calls should be delivered with the DID (aka DNIS, DDI, etc). Usually
you pick this up as the ${EXTEN} in your dialplan and go from there.
[snip]
Broadvoice, however, won't let me change the outgoing caller ID.
Apparently they have to do this on a trunk by trunk basis. So if I want
to have an outgoing call go through line 1 (let's say its ACME Inc), I
want it to show 'XXX-XXX- Acme Inc' for the Caller ID.
[snip]
Being able to specify the caller ID number depends on the carrier.
Vitelity.net does. Specifying the caller ID name is not going to work. The
way it works (from 40,000 feet) is that the name is not passed onto the
real telephone system. The carrier for the dialed number looks up the
number in a database and presents that to the dialed number. If you dial
another VOIP account (sip:john-sm...@example.com) your caller ID name
should be passed.
Does this sound right? Should I have purchased all separate trunks up
front and then have the phone number transfer associated with the trunk
for it? Or is this only something that will affect outgoing calls, so
its not a big deal? And what about when the line is busy? How is that
handled? I was on the phone yesterday when another call came in, and it
came in, jumped to a different extension and then eventually went to
voice mail as I didn't answer it. Will my plan to use one trunk for all
incoming lines make sense here, or am I likely to get all of this mixed
up with calls coming in for one business and being routed to the wrong
place?
I'm more comfortable with the word account than trunk. You can have
multiple DIDs numbers associated with the same account. Some providers
make you specify (via their web site) where you want the calls to go. Some
make you configure your Asterisk server so it registers with their
server. I prefer registration because it let's me change things around
easier.
I had this issue with Teliax. Basically with SIP, Teliax could not (or
the protocol won't let you) set your outbound caller ID via Asterisk.
Caller ID is set on a per account basis with Teliax when using SIP(IAX
was not working well for me with Teliax). So I have two outbound pay per
minute accounts with them. One for our home use and one for my business.
I use 51 prefix for home outbound calls and 52 prefix for business
outbound calls. Then my dialplan selects the proper account at Teliax
and you get the proper caller id set.
My inbound is still pots lines from the telco, btw. There is no
significant cost savings on inbound for telco vs VoIP here.
Lyle Giese
LCR Computer Services, Inc.
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Re: [asterisk-users] OT - Do analog gateways detect a phone is plugged in or out ?
Philipp Kempgen wrote: Steve Totaro schrieb: On Wed, Jul 22, 2009 at 12:07 PM, Steve Underwood ste...@coppice.orgwrote: Olivier wrote: I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send qualify queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Unless there is a call in progress the switch has no idea what phones might be plugged or unplugged. Nothing happens on the line what it could detect. It certainly would seem possible and would be a great feature request. There probably is no circuitry existing to do it, but I would assume that ohms, volts, or something could be measured while sending a small amount of voltage down the FXS lines. Bonus point will be given for detecting the phone model and color as well. ;-) Philipp Kempgen Yes, it's technically possible for the phone company to determine if there is a set or something connected to a phone line. It involved hitting the line with +130v dc test voltage and reversing it quickly and seeing how much capacatance kick there is. This kind of testing is normal for telco CO lines. FXS chan units or gateways normally do NOT have this built into them. The only exception I know about is SLC(Subscriber Line Concentrator, which is a generiac term for fiber or digital lines feed to telco boxes in the field). And even there the process was to have a cut-in relay and connect the out cable pair back to the CO via a dedicated copper pair to do these tests via a device called a PairGain Test Controller. I know because I was an 'expert' on them and traveled around going from telco CO to CO fixing them. In other words, there is some circuitry involved in doing these tests and I don't see any PBX, FXS chan unit or gateway manufacturer rushing to add more to this to their product line. They have not done it yet and I don't see anyone other than the phone company willing to spend the money to make it happen. To keep this on topic for Philipp's remark, the only bonus points we assigned was to correctly guess how many phones were attached to the phone lineGRIN! Lyle Giese LCR Computer Services, Inc. ___ -- 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] tdm loosing interrupts and latency
Alex Samad wrote: Hi I have come across a problem, with my tdp410 and soekris board (basically pc on a chip amd geode cpu). I am using the box as a firewall/asterisk box. The problem occurs when I drop ppp and I get dead loop dectiotn going, I seem to lose interrupts and get lots of messages in syslog from wctdm24xx saying missed interrupt increasing latency its out lined here (http://forums.digium.com/viewtopic.php?p=126997highlight=sid=9de59f41f1a93ee8701b28fdd0cf6073) Seems like the driver (and this is in zaptel dadhi code), increases latency by +1 until 30. and then the card seems to not work. In my case I have seen latency increase from 8m (I have this as a starting point in the module load) up to 17ms usually around here the fxs and fxo ports stop working . I have to unload and then reload the module. bummer. I can think of a couple of solutions 1) build some intelligence to bring down the number when things are okay 2) build logic to say if a number is provided on module load to fix it to that 3) add a sysfs (/proc) interface to allow changing this value on the fly I could also try and solve my problem with the dead loop detection cat /proc/interrupts CPU0 0: 23809265XT-PIC-XTtimer 1: 0XT-PIC-XTi8042 2: 0XT-PIC-XTcascade 4:255XT-PIC-XTserial 5: 459544XT-PIC-XTeth1 8: 0XT-PIC-XTrtc0 10: 95177163XT-PIC-XTwctdm24xxp0 11: 28938443XT-PIC-XTeth0 12: 28938632XT-PIC-XTeth3 14:3624228XT-PIC-XTide0 15: 1XT-PIC-XTehci_hcd:usb1, ohci_hcd:usb2 NMI: 0 Non-maskable interrupts LOC: 0 Local timer interrupts TRM: 0 Thermal event interrupts SPU: 0 Spurious interrupts ERR: 0 MIS: 0 as you can see with the interrupts the wctdm24xxp0 is above eth0 (local lan) and eth3 (my adsl) eth1 is wireless and not heavily used So any one had this problems, any other possible solution to this ? How to engage digium to providing a fix for this ? Alex If your ppp is dropping, that means you have lost Internet connectivity, correct? If that is the case, then that is your problem as Asterisk does not tolerate the lose of DNS resolution very well. Lyle Giese LCR Computer Services, Inc. ___ -- 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] Maximum cable length for analog phone from FXS port
asterisk-us...@rogg.is wrote: Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin It's not expressed in distance. They will supply the current voltage output and you need to apply ohm's law. That requires knowing the resistance of the cable which is dependent on length and gauge. Lyle Giese LCR Computer Services, Inc. ___ -- 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] Maximum cable length for analog phone from FXS port
Even with 'conventional' PBXs, there is such a thing as power fail devices where the extension is cut to a telco pots line for dial tone if the PBX goes down. Jon Pounder wrote: John Novack wrote: If this is an emergency phone situation then I would question the wisdom of even considering using Asterisk. Conventional telephony solutions exist that will easily cover the loop length and provide the reliability that should be required by risk management in such a situation. why are you going on the assumption asterisk is somehow inherently less reliable than a conventional solution ? I am not trying to start any sort of war here, but is that based on any sort of facts ? hardware wise its basically all the same electronics whether they were meant as a general purpose computer or a telephony specific computer - they all fail eventually and the MTBF is usually related to the relative price in the specific market. I have not really had any software reliability problems in years of running asterisk (although some do and I am sure there are firmware revs for pbx's that have issues too) so why make that general statement ? as far as risk management - any one system can fail, end of story. Risk management would entail a backup system if failure of the primary is not acceptable. In a tunnel application physical damage to the wiring is probably a lot more likely than a hardware failure, be it from accident, fire, collapse etc., meaning when you need the phone most, it is least likely to work. Those factors would affect any hardwired telephony solution equally. John Novack asterisk-us...@rogg.is wrote: Appreciate all your input folks. Much of it very helpful in the greater context of the initial question. Thank you for the suggestion of using various wireless devices, but I'm stuck with fixed wiring since this is a security/emergency phone(s) installation underground in large tunnels. Also, switching to VOIP is not really the answer here because then I'm forced to solve a lot of power, repeaters/switches problems that arise. So I'm actually worse of than using the analog connections I think. I do have some control over the wiring/cable chosen for this project but still forced to find a solution where I can feed the analog phone line the total 3km line distance. I would love to find a way to do this in the Asterisk context with some sort of FXS feed, either from Digium (or compatible) hardware or any of the available ATA boxes. The Sapura box suggestion may be something and I'll look closer into that as well as continuing to look for other ways to do this. tnx! Baldvin -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: 26. maí 2009 19:42 To: novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS port I would suggest making a wifi connection with directional hi-gain antenna's. Ans a small box at the other end. Have a look at: http://www.fit-pc.net/fitpc-2-p-2.html or http://www.fit- pc.info/downloads/handleidingen/fit_pc_2_eng.pdf ___ -- 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: 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] PSTN Connection
Brent Vrieze wrote: Lyle Giese wrote: Manoj Panicker - FOES wrote: Hi Which is the best interface card to connect* PSTN* line with Asterisk. Can somebody please help. My intention is to route the incoming PSTN calls to internal IP Phones through Asterisk and Vice versa. The Asterisk is in LAN and is reachable from all the IP phones in the LAN. Thanks Manoj That's a wide open question. How many lines? What kind of lines? What country are you in? What options are availible to you? I only have three incoming lines for a soho Asterisk install. I decided on a T1 card and picked up a used channel bank on ebay. Not the cheapest way, but it has served me very well. You are not going to get much help unless you define the problem better. Lyle Giese LCR Computer Services, Inc. HI, OK, I'm going to chime in on this one as I am going to set up an Asterisk system for our volunteer ambulance service. As a part of the Emergency Services we need to maintain a POTS line as redundancy and due to the fact that with an old style phone I don't need power for the phone to work. I plan on using a SIP provider for the rest of our phone needs. If not for the emergency services part I would go completely SIP based. Anyway I would need a FXO/FXS card for use in the US. Only one line so I don't need any of the fancy 4 line systems. I have heard you can use certain modems to do this but I would like what I am doing to be seamless and not require hacking at a problem for hours to save $50. I just want it to work quick and easy. I am unsure what you mean by What kind of lines? and What options are availible to you?. Maybe that is part of asking this question, to get some info about the phone system too. Any help would be grand. Thanks Brent ___ -- 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 Brent You have defined what you are going to do, basically a small system and only need one POTS line. You could also use an ATA to convert a POTS to SIP to go into the Asterisk box. That would probably be a more supportable solution as those devices don't appear to be disappearing off the market like that modem solution is. Then if in a couple of years, lightening takes out the converter, you have a purchasable solution. You can also do this with Digium cards. Lyle ___ -- 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] PSTN Connection
Manoj Panicker - FOES wrote: Hi Which is the best interface card to connect* PSTN* line with Asterisk. Can somebody please help. My intention is to route the incoming PSTN calls to internal IP Phones through Asterisk and Vice versa. The Asterisk is in LAN and is reachable from all the IP phones in the LAN. Thanks Manoj That's a wide open question. How many lines? What kind of lines? What country are you in? What options are availible to you? I only have three incoming lines for a soho Asterisk install. I decided on a T1 card and picked up a used channel bank on ebay. Not the cheapest way, but it has served me very well. You are not going to get much help unless you define the problem better. Lyle Giese LCR Computer Services, Inc. ___ -- 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] Parked Calls Problem
Brent Vrieze wrote: openSuse 11 Asterisk 1.4.23.1 Asterisk GUI 2.0 When parking a call it does not tell me what extension it parked the call on. I think I read something in the mail list that mentioned a problem with call parking and one of the Asterisk 1.4s. Is 1.4.23.1 one of those version having issues? Thanks I have some Grandstreams and a couple of Uniden phones. The Grandstream TRNF button does an un-attended transfer and does not get the announcement. The Uniden does an attended transfer and gets the announcement. This is on 1.4.13 Lyle ___ -- 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] Sangoma A104d and Adtran 850 problems
A channel bank != PRI. A PRI is ISDN. A channel bank is not the same as a Primary rate ISDN line. With a channel bank, each channel's signaling is done in the channel. Primary rate ISDN has a D channel to contain all signalling for the 23 voice channels, taking over the 24th voice channel. Lyle Jim Dickenson wrote: I have a system that I am trying to get a port on a Sangoma A104d card connected to an Adtran 850 with 5 FXS modules and 1 FXO module. A problem I am having is figuring out what cable should be used from the port on the Sangoma to the JP2 port on the Adtran. Tried was a cross-over T1 (1-4, 2-5, 4-1, 5-2) as well as a straight T1 (1-1, 2-2, 4-4, 5-5). Neither one made the Sangoma port show a green light, only red. Also the best I can tell the Sangoma port gets configured the same when connecting a PRI line or a cable to the channel bank. Is this correct? It is /etc/dahdi/system.conf that says what is connected to the port, correct? Here is /etc/wanpipe/wanpipe7.conf: [wanpipe7] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 9 FE_MEDIA= T1 FE_LCODE= B8ZS FE_FRAME= ESF FE_LINE= 3 TE_CLOCK = NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 0DB FE_TXTRISTATE= NO MTU = 1500 UDPPORT = 9000 TTL= 255 IGNORE_FRONT_END = NO TDMV_SPAN= 7 TDMV_DCHAN= 0 TDMV_HW_DTMF= YES [w7g1] ACTIVE_CH= ALL TDMV_ECHO_OFF= NO TDMV_HWEC= YES Here is /etc/dahdi/system.conf loadzone=us defaultzone=us #Sangoma A104 port 3 [slot:4 bus:9 span:7] wanpipe7 span=7,0,0,esf,b8zs fxols=145-164 fxsls=165-168 Here is /etc/asterisk/chan_dahdi.conf: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A104 port 3 [slot:4 bus:9 span:7] wanpipe7 context=to-cbfxs group=2 echocancel=no signalling=fxo_ls channel = 145-164 context=from-cbfxo group=3 echocancel=no signalling=fxs_ls channel = 165-168 Held figuring this out would be appreciated! ___ -- 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] FXO Ignore ring
Cary Fitch wrote: Is there a way to program an FXO device to totally ignore incoming calls? I want to put an FXO on a Fax line so that 911 calls can be sent via that line, but all other activity on the line is between the Fax machine and the phone company. Perhaps munge the ring tone detect if nothing else? Cary this works here in my extensions.conf(with my fax line in this context): [outonly] exten = s,1,Wait,20 ; setup for fax line to stop ringing exten = s,2,Hangup Lyle Giese LCR Computer Services, Inc. ___ -- 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] Eyebeam or Xlite
David @ULC wrote: Lets presume that my both software are open. Xlute and Eyebeam But I want my calls from Asterisk to land only on Eyebeam and Not on xlite. How to set it ? Give each their own SIP credentials. Then in Extensions.conf, when dialing into your extension, send the call to both SIP devices. Then both will ring on your computer and you can decide which to answer. Caution, I have not tested this scenerio, but it should work as long as the two applications are not trying to use the same orginating port numbers to contact your * server. Lyle ___ -- 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] CentOS and BAT File
David @ULC wrote: In windows, we use BAT file to execute few series of command , which help us in not writing each command manually everytime we want to execute those commands. In CentOS, I want to do the same thing. Any Advice ? ___ -- 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 They are called shell scripts. ___ -- 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] Ntework Card
Why? This is not an Asterisk problem... You need to find a forum specific to your linux distro... Lyle David @ULC wrote: Sorry to bump it , but any help ? Like un-installing the driver and reinstalling it will solve the issue ? Or shld I reinstall the OS again ? On Sun, Jan 25, 2009 at 2:53 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: *Quote:* [r...@vicidialnow src]# service network restart Shutting down interface eth0: [ OK ] Shutting down loopback interface: [ OK ] Bringing up loopback interface: [ OK ] Bringing up interface eth0: [ OK ] Bringing up interface eth1: skge device eth1 does not seem to be present, delaying initialization. [FAILED] [r...@vicidialnow src]# eth1 is the Oboard card. I did install driver but its same. I am sure Driver is a correct one as after installing driver it showed me the card which is eth1 but due to soem issues I deleted that network interface from webmin. But now its NOT detecting. Kindly advice ___ -- 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] Suggestions on how to create a hunt or hunt like (rollover, multi-line) group or where to get one?
How many incoming calls will they support per line? You may find that they support more than one incoming call per number. Otherwise, get another provider. Lyle Alfred Monticello wrote: I'm still stuck with this problem..Would appreciate any ideas anyone might have on this one. Thank you *From:* Alfred Monticello ajmce...@yahoo.com *To:* asterisk-users@lists.digium.com *Sent:* Monday, January 19, 2009 12:09:12 PM *Subject:* [asterisk-users] Suggestions on how to create a hunt or hunt like (rollover, multi-line) group or where to get one? I have about 5 incoming USA SIP lines, but my provider does not have any sort of roll-over or huntgroup feature. Does anybody have an idea on how I can create a general number that will ring to the next available, non-busy SIP line that I have? Is there a provider out there that would do this? Any suggestions would be greatly welcome. Thank you. ___ -- 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] Description of Zaptel/DAHDI E1 alarms
Lukas Rypl wrote: Hello, I am missing any description of zaptel/DAHDI alarms. The TE200 series user manual contains only a description of LEDs states. These alarms states are visible in zttool/dahditool or in astersick CLI (zap show status) and I wonder what is the real meaning of these alarms for E1 channel. Possible alarm states (based on zaptel.h 1.2): 1. No alarms 2. Recovering from alarm 3. In loopback (local loopback or far end?) 4. Yellow Alarm (is it only Far end Loss of Frame?) 5. Red Alarm (Loss of Signal?) 6. Blue Alarm (AIS?) 7. Not Open Thank you for any help. Lukas Rypl ___ -- 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 Notes: 3 In loopback means I have been asked locally to provide a loopback of some sort to somebody. 4 Yellow means the far end does not like the signal received for what ever reason and the far end is trying to tell me the circuit is broken. 5 Red means I don't like the signal received. Could be framing issues, could be CRC errors, could be no signal, could be ? 6 Blue alarm means I am receiving AIS or all ones signal, can be framed or unframed. Lyle Giese LCR Computer Services, Inc. ___ -- 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] caller ID - handle_request_invite: Failed to authenticate user
Joseph wrote: We have a caller ID from our phone provider Shaw Cable (digital phone) and it was working OK until recently. I get an error: WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have 4, digest has pstn- NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user THELMA sip:7804789...@10.10.0.103;tag=50e17675d59121c4o1 at this point call fails, it is not being passed through to asterisk. I'm using Linksys 3102, PSTN answer delay is set to 3sec. to allow for caller ID to pass through. When I decrease timing to 1sec. or eliminate it 0sec the call goes through but there is no caller ID being forwarded. It was working OK for a while. So I'm not sure if Shaw Cable have upgraded something on their digital phone or there is a problem with asterisk/ 4 is a Line1 pstn- is PSTN Line Have you tried to extend that delay to 5 or 6 seconds? It's possible that caller id is being sent a second or two later/longer, but your 3 seconds is now cutting off a portion of the caller id data. Lyle ___ -- 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] how to debug mime-construct with fax2mail?
If you are running the script within Asterisk as root, then it's a path environment issue. My guess(and I run into this with cron jobs all the time) is that the path is different from the command line than the environment that the script runs under. There are times where the fix is to use the fully qualified path when calling stuff and not assume it's in the path. Lyle sean darcy wrote: Joseph L. Casale wrote: Have you tried your system stuff under su - asterisk? Once it works that way, the system() command will work. asterisk is running as root, I run the command at the terminal as root. I am guessing he doesn't even have an asterisk user. Well I do have an asterisk user, and once spent a weekend trying to run asterisk as asterisk user. But I don't see what this has to do with my problem. The System() cmd works: I can see the log from fax2mail showing it was called, and called with the arguments I expected. So System() did it's thing. What I can't figure what is why fax2mail really works from the command line, but fails to effectively call mime-construct when called from System(). I was hoping someone who has used mime-construct could show me how to debug it. It may be a permissions problem, but since both run as root it seems unlikely. In any event, being able to debug mime-construct would allow me to figure it out. sean ___ -- 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] Not Dialing 9
Gordon Henderson wrote: On Thu, 8 Jan 2009, Thczv F. Thczv wrote: When I set up my Asterisk box at home I didn't want to have to dial 9 to dial off premises, so I gave all my local phones three digit extensions with this format: 1[1,0]*. My thought is that there are no area codes that start with 0 or 1, so if I use those numbers, I can create 20 local extensions that can be dialed with 3 digits, and not have to use a timeout when dialing long distance. If I dial 1, then anything other than 0 or 1, Asterisk knows I am dialing long distance. If I start with any number other than 1, Asterisk knows I am dialing a local or local toll call. This has worked fine for me (as far as I know). Is there some flaw I am not seeing? I see a lot of small businesses that require a 9 to dial out, even though they don't have very many extensions. Couldn't they do what I did and not have to dial 9? I ask because we are having a problem where I work with our Cisco 7940 phones adding an extra 1 sometimes, which gets the local Sheriff upset (too many 911 calls). You don't say, but I'm guessing you'r in the US, or at least not Europe. Starting extensions with 1 isn't a good idea in Europe, as our equivalent of 911 is 112 (and 999 in the UK) Gordon The norm in the US is going to 8 instead of 9 for the outside line. I use 8 and still use 86 for voice mail(vm). But using 1 or 0 like you suggest could cause problems with international dialing. Lyle ___ -- 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] [SPAM] enabling silence suppression in asterisk
bala krishnan wrote: Hi Friends, Currently i am using the asterisk 1.4.x version. In that i want to enable to silence suppression in the SIP calls. Please tell me the configuration changes to be done. Thanks in advance, balasam. Enabling silence suppression is a bad thing. Asterisk and sip phone will think that the other party has dropped off and will randomly drop calls on you because of a lack of traffic from the other party. Lyle ___ -- 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] Meetme - play the name
sasikala kala wrote: Hi, I have a requirement, whenever a user comes into the conference, it has to announce the user name to all the person who are all available in the conference. I have used Meetme(,di) where i is to announce the user leave/join with review. I user used I also, which is to announce the user leave/join with out review. In both the above cases, it is prompting the user to say their name, but what i want is, if it gets the name one time, thats all, it should just play that name whenever the call comes from the same callerid. That's not a realistic expectation. How can you presume that because callerid is xyz that it's always the same person calling? You can not. Office's routinely have one main number with callerid being the same for all office users. I would not be surprised to find two users calling in separately from the same office having the same callerid, where you can not tell them apart based on callerid. Now having said that. I can see where you could/would be able to get the name announced once on arrival and get meetme to save that and announce they left and then forget the recorded name. If they were disconnected, they may need to re-record their name. I don't know if that is a feature now or not, but that would be doable and you could ask for a new feature based on this description. Lyle Is it possible to achieve this feature by some way? Hope somebody would have the same requirement, kindly help to achieve to do the same. thanks and regards Sasikala. Add more friends to your messenger and enjoy! Invite them now. http://in.rd.yahoo.com/tagline_messenger_6/*http://messenger.yahoo.com/invite/ ___ -- 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] Sendmail for Voicemail
You need to implement SMTP-AUTH and log in when sending mail to your smart host. I have a template for Postfix to do that. Many *nix distros have Postfix with a sendmail compatible binary in front of it. Lyle Giese LCR Computer Services, Inc. [EMAIL PROTECTED] wrote: When I send email from my local asterisk machine, my IP address get's RBL'd. Asterisk is my only reason for running sendmail, so to keep it simple, I tried to make my ISP's mail server a 'smart host' (relaying to a trusted mail server) but my ISP doesn't allow ANY kind of relaying these days. I imagine there are many like me who are not sendmail experts who want to send Asterisk Voicemal. Can someone direct me to the quick, dirty and secure way to send mail from my asterisk box? The good news is that I'm on a Fixed IP on a registered network with working reverse in-addr.arpa lookups, and as you might have guessed, all mail would originate from the local host. Suggestions? Thanks! -Karl ___ -- 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] t1 cards
T1 is NOT DSL. Most T1 links you purchase now are brought into your building with a type of DSL conversion to extend the distance between repeaters/amplifiers. T1 is purely a digital signal. DSL converts the ones and zeros to audio(multiple tones to provide multi channels of data). A simple analogy is comparing a T1 to DSL as a serial port to a modem. Back in the old days before fiber, copper T1's between CO's had their repeaters placed aproximately 1 mile apart. Best case going T1 port to T1 port, I would not expect this to work reliably at distances greater than one mile or 1.6 km but that does depend on the quality of the cable also. But in my mind, I would be seriously concerned about lightening protection. I have been around telco's and privately owned facilities for a long time and see lightening to be a very serious issue in this scenerio. I have seen short distance copper replaced by fiber because of issues over time with lightening damage despite having proper telco grade protection. Lyle Jeff LaCoursiere wrote: I would say miles. DSL limits for equiv bandwidth is around 3 miles if I recall correctly. j On Fri, 3 Oct 2008, Eric Fort wrote: without any other hardware than 2 bare ass pci based t1/e1 cards wired back to back how far can one go between them? additional hardware defeats the purpose. Eric On Fri, Oct 3, 2008 at 3:01 AM, Gordon Henderson [EMAIL PROTECTED][EMAIL PROTECTED] wrote: On Fri, 3 Oct 2008, Eric Fort wrote: yes, more than 300 meters (longer than copper based ethernet allows). Yes to E1, as I understand it, it's just a config change on many cards anyway. I'm specificly looking at pci based t1/e1 cards because I'm finding single port cards on ebay going for 100-200 usd. in some cases I may want to drive a channel bank at the far end, thus t1/e1. anyone have experience on how far these pci based cards will drive when wired back to back? Looks like this is the thing then: http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468mid=5362 Just over $1000 a pair... couple that with an OpenVox PRI card at one end, channel bank at the other, and off you go... Gordon Eric On Thu, Oct 2, 2008 at 11:34 PM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thu, 2 Oct 2008, Eric Fort wrote: I presently need to connect a few channels of voice and data between multiple locations where I own the copper between them. Each location exceeds 300M from any other location. I'm thinking of generating T1's and running those between locations. If I use PC based cards wired back to back (I can do that, right?) what kind of distance can I expect to be able to span without needing repeaters? What inexpensive cards can you recommend for use with asterisk? I'm considering either digium or sangoma. Would I get any better performance if I used a sync-serial card connected to a separate csu/dsu? 300 metres, right? (not 300 miles?) Why stop at T1? Go for E1 :) with the right kit at each end you ought to be able to get 2Mb/sec or more. (distance depending) Personally, I'd go for a technology that gave me Ethernet at each end - then it makes it much easier to mix voice and data - But using something like a sync. modem and line driver then you need a media converter of some sorts at each end which might bump up the cost - at the savings of the E1 card in the PC though. Last time I had bare copper to play with (a BT EPS8 circuit) I had a 2Mb modem at each end going into a Cisco 2600 which was running CHDLC over the link and acting as nothing more than a dumb media converter to give me Ethernet at each end. This was 6 years ago though. Ah, Looks like the technology has improved somewhat: http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468mid=5261 From the UK site: Or even: http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424mid=4946 (same thing from the UK site:) http://www.blackbox.co.uk/solutions/display.asp?cs=dvhid=1doc=lb300a-r2tx=LANsx=Network%20Appliances You need a pair, obviously... Hm. US site is $305, UK ?253. Rip-off Britain again by the looks of it As for inexpensive cards - OpenVox. Their E1 cards seem to work OK, but if using a LAN extender, then they're not neeed at all... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE
Re: [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting
Brian J. Murrell wrote: I'm looking into getting a new phone and wondering what the difference in functionality is between a single line phone with call waiting and a real 2 line phone (either a real SIP phone or an analog 2 line phone and a 2 port ATA) is. Why would I want the real 2 lines vs. just being able to take an incoming call via call-waiting? Cheers, b. 1) a two line phone can register with two different * servers or sip carriers. 2) It's easy for both incoming and outgoing to separate business from personal calls. (ie line1 is personal, line2 is business) 3) It's easy for a two line phone to register to two different accounts on * and then subsubscribe to two different MWI's on different VM boxes(again goes back to seperating business from personal or your VM from your significate other's VM) That's just off the top of my head. Lyle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting
Brian J. Murrell wrote: On Tue, 2008-09-30 at 08:23 -0500, Lyle Giese wrote: 1) a two line phone can register with two different * servers or sip carriers. Indeed. But if I only had the one * server which itself registered to my carriers... 2) It's easy for both incoming and outgoing to separate business from personal calls. (ie line1 is personal, line2 is business) Yeah. Given this is a home office phone though, that I even route the house calls to it is just a convenience for when I am in the home office. IOW, if I'm in the office, I almost always want to answer it vs. if I am at a personal/house phone, indeed, I may not want to answer business calls, but this is not the case... 3) It's easy for a two line phone to register to two different accounts on * and then subsubscribe to two different MWI's on different VM boxes Ahhh. Now this is an interesting possibility. (again goes back to seperating business from personal or your VM from your significate other's VM) Ahhh. Indeed. This use case is worth considering. Although, really, I want to migrate to VM in IMAP so that I don't even (have to) use the phone to know there is VM or listen to/delete it. I would use my e-mail client which is my preferred interface. I have never been convinced that VM via email is a convenence. You have to use the loudspeakers on the PC or headphones, which is not as convenient as a handset. Not to mention the privacy issues/problems using loudspeakers for VM. Do you want your kids/wife overhearing your customer that is upset with you? I find that the email notification is more than enough to know who called and many times why without listening to the actual message and deciding how urgent it is to listen to the message or deal with it. In any case, this one is an interesting benefit. Not sure I'm convinced enough yet though. That said, thanks for the input Lyle, I really appreciate your thoughts on that. b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] Asterisk and Network Monitoring
Dean Collins wrote: Has anyone ever 'released' an Asterisk module that is easily shared/downloadable? Or doesn't the nagios open source code work like that? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Tuesday, 9 September 2008 9:29 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk and Network Monitoring On 14:50, Tue 09 Sep 08, Jacobus van Niekerk wrote: Dear Asterisk Users I'm looking for a solution that can be used to monitor Asterisk and the Telco lines aswell as the network (Servers, WAN LAN links, Router Switches) We use nagios for that. If you use an asterisk module to monitor the health of asterisk and the server both are running on crashes/loses power/overheats, what server/service is going to send you the notification that it's down? You put the monitoring software on a different cpu for a very good and valid reason. Lyle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] 5 min limitation on phone calls! how to!
RoLaNd RoLaNd wrote: Hello all! my last month's phone bill sky rocketed after i setup asterisk with softphones all over the house! could someone help me set up a limitation for my wife and kids not to be able to talk for more than 5 min at a time! or like 20 min per week! or whtever limitation i could set for this! any help would trull be appreciated:) I would check the CDR logs first and make sure that a hacker did not get into your * box and is making calls on your dime. Lyle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] email notification to external email address
Brian Simpson wrote: All, I have a problem. The company I work for has been bought out by a bigger company. The employees are in the process of changing all their email addresses to the new company name. I have my voicemail.conf file setup to email users when they have a voicemail message. The mail server that was used to notify everybody is on the same network as the Asterisk PBX. Now I have to change all the emails in the voicemail.conf file to the new company's email addresses. The email server for them are external of the network that the Asterisk sits on. I have change a couple to test but the email notification is not happening. Any idea what is going on and how to resolve. Is there something else that I need to do to get the emails to work? I am new to the Asterisk and have been forced to take over for someone that has left the company. I do have telephony experience with Legacy systems. Any help is appreciated. Most likely the box is using sendmail or postfix to send those emails out. You need to setup sendmail/postfix to use a smarthost using smtp auth to allow relaying from this box. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] Call Recordings...
I bet the reason is that when his gf calls, he can erase the records so his wife's divorce attorney can not get his hands on them to play in court. Lyle Eugen Soare wrote: So basically, He wants all calls recorded, but he wants a sequence that he can push, so that when he rants and raves at a customer, there won't be evidence to say that he did that... :) Just a hunch on that. :) I don't know. Eugen On 7/22/08, *Gregory Malsack* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, My boss is asking me to setup the asterisk server to record all calls. (Simple). However, he wants to be able to enter a key sequence during the call to stop the recording. Any ideas on how I would do that? Thanks, Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] problem in making call pc to phone vice versa
Your E1 links are down. (red alarm) Your card does not like or see your providers E1. Lyle Bikrish Amatya wrote: Hello everybody I have configures asterisk server and i am using TE220P digium card. Here is the content of the /etc/zaptel.conf file ### span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,2,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 loadzone= in defaultzone = in the content of /etc/asterisk/zapata.conf is as follow [channels] context=incoming switchtype=national ;pridialplan=national usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no callprogress=no callerid=asreceived group=1 channel=1-15,17-31 # output of zttool is as follow #9474; Alarms Span #9474; #9474; RED T2XXP (PCI) Card 0 Span 1 #9474; OK T2XXP (PCI) Card 0 Span 2 #9474; Output of cat /prox/zaptel/1 is as follow Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 HDB3/CCS RED 1 TE2/0/1/1 Clear (In use) RED 2 TE2/0/1/2 Clear (In use) RED 3 TE2/0/1/3 Clear (In use) RED 4 TE2/0/1/4 Clear (In use) RED 5 TE2/0/1/5 Clear (In use) RED 6 TE2/0/1/6 Clear (In use) RED 7 TE2/0/1/7 Clear (In use) RED 8 TE2/0/1/8 Clear (In use) RED 9 TE2/0/1/9 Clear (In use) RED 10 TE2/0/1/10 Clear (In use) RED 11 TE2/0/1/11 Clear (In use) RED 12 TE2/0/1/12 Clear (In use) RED 13 TE2/0/1/13 Clear (In use) RED 14 TE2/0/1/14 Clear (In use) RED 15 TE2/0/1/15 Clear (In use) RED 16 TE2/0/1/16 HDLCFCS (In use) RED 17 TE2/0/1/17 Clear (In use) RED 18 TE2/0/1/18 Clear (In use) RED 19 TE2/0/1/19 Clear (In use) RED 20 TE2/0/1/20 Clear (In use) RED 21 TE2/0/1/21 Clear (In use) RED 22 TE2/0/1/22 Clear (In use) RED 23 TE2/0/1/23 Clear (In use) RED 24 TE2/0/1/24 Clear (In use) RED 25 TE2/0/1/25 Clear (In use) RED 26 TE2/0/1/26 Clear (In use) RED 27 TE2/0/1/27 Clear (In use) RED 28 TE2/0/1/28 Clear (In use) RED 29 TE2/0/1/29 Clear (In use) RED 30 TE2/0/1/30 Clear (In use) RED 31 TE2/0/1/31 Clear (In use) RED I am new to asterisk and googled around , configured the asterisk server. Now when i make a call from outside , it give me busy tone.. and when i call from softphone .. it shows me as show below -- Executing [EMAIL PROTECTED]:1] Dial(SIP/bikrish-09b21980, Zap/g1/600833) in new stack [Jul 3 19:14:34] WARNING[6018]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/bikrish-09b21980' status is 'CONGESTION' I am not able to figure out the problem. Any kind of help would be appericiated. Thanking you bikrish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] Recommendations for Motel Instalation.
Jay R. Ashworth wrote: On Fri, Jun 20, 2008 at 03:42:28PM -0600, Arturo Ochoa wrote: I have a customer who owns a little Motel, and he wants to upgrade to a Asterisk PBX. There is one analog phone per room (aprox 80), and the cable is CAT 3. You might want to consider snagging an FXS channelbank off of eBay (we use the Zhones, which work pretty well for us), and using a multi-port T-1 card. If this is not a business motel, you'll likely get by with 24 trunks, so a quad-T card would support both your incoming lines and 3 channel banks (we seem to pay about $180-240 for them, making this cost effective), assuming approximately 80 isn't more than 72. :-) If that's not enough ports, then yeah, you'll probably be best served going to a Ethernet gateway; I personally have never liked the idea of stuffing that much FXS inside a PC chassis. Cheers, -- jra I agree with using used chan banks off of Ebay, but I would not touch a Zhone. I had one and sold it as soon as I could. They are a real PITA to program and don't pass caller id. I have purchased several Adtran chan banks and have been extremely happy with them and tech support at Adtran. Support from Adtran has been nothing short of excellent even though they knew I calling about used chan banks from purchased on Ebay. One note is if the admin/craft interface has a password on it, you have to call Adtran to reset it. There is a way to bring up a numeric challenge code and support will tell you the response to it and you are in. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
Tilghman Lesher wrote: On Thursday 12 June 2008 03:23:46 Mark Adams wrote: I appreciate the responses thus far but I am looking to find out what type of security I should implement for the future. Being new to linux, not to mention asterisk I didn't realize that someone could brute force into the box and upload crap. With that in mind it seems that I would want to get a hardware firewall such as a hotbrick or a sonicwall firewall. One of the most frequent security issues comes not in the form of a software flaw, but simply in people choosing easy-to-guess passwords on the root account. There are two suggestions I have to reduce the risk of this brute force. First, choose a username that is uncommon. In your case, do not use 'root', 'admin', or even 'mark'. 'madams' might be a good choice. Once you figure out that username, configure sshd with the AllowUsers directive to ONLY allow logins from that user. If you need root access, install sudo. If an attacker cannot figure out what your username is, then it doesn't matter even if they guess your password, because they aren't getting in. And of course, the second part is choosing a secure password, one that contains mixed case, numbers, letters, and symbols. Don't be afraid to write down that secure password, as long as you keep it on your person (wallet is a good choice). 99% of the attackers who might otherwise compromise your machine will never come within 1000 miles of you. However, your wallet contains things that are far more valuable than your password (your identity documents, for example), so it is hoped that you will be able to keep that password away from people who would otherwise do you harm. Most recent hacks that I have first or second hand knowledge of came from ssh issues. Most inexperienced admins will expose ssh without using the 'allowgroups' option in their sshd_config and will get hacked by someone logging in via ssh using a system account with no password. The second thing to do with ssh is to move it to another port to keep the script kiddies from pounding on it. If there is a weak or missing password, they will find it. An encrypted USB thumbdrive is also a good storage device for passwords. I use TrueCrypt and have the executable availble unencrypted on the thumbdrive so I could plug it into almost any machine and get to the encrypted data. Lyle Giese LCR Computer Services, Inc. ___ -- 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] remote server with Snom 190
Ronald Wiplinger wrote: I have a local asterisk 1.2 and a remote asterisk 1.4. Snom 190 can be used with the local asterisk but not with the remote one. I need some hints where to track down this issue. Some information: Snom 190: Line 1: Account: 615 Password: OnlyIknowit Registrar: ast.mydomain.com Status: OK Line 2: Account: 6888 Password: Otherside Registrar: 22.33.44.55 (only IP address!) Status: Not found Function keys: P1 Line Number sip:[EMAIL PROTECTED];user=phone P2 Line Number sip:[EMAIL PROTECTED];user=phone Remote server is a fresh installed Ubuntu 8.04 server. What do I miss? bye Ronald ___ -- 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 NAT issues? Is the remote server on a private IP address behind a NAT firewall/router? Firewall issues at either end? At the appropiate ports open on both firewalls for the phone to talk to the remote Asterisk server? Lyle ___ -- 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] Error Counters on PRI Circuit
Jay R. Ashworth wrote: On Tue, May 20, 2008 at 07:03:06PM -0500, Lyle Giese wrote: Is there a way to see error counts on the T1 of a PRI? Hooked up to asterisk via a digium TE122. Looking for something to make sure I'm not getting any CRC, framing or other errors on the T1. Go on ebay and buy an ADC Kentrox DataSmart 658 for less than $100 dollars. The 658 has an ethernet port for management and grabbing the stats of the T1 line itself. For an ISDN T1 PRI, set channels 1-23 to T1 Voice and Channel 24 to T1 Data, otherwise the B channel won't come up. You may need a DB15male and a DB15female to RJ45 adapters as not all of these units on Ebay come with them. The T1 from the phone company connects to the Network connector(db15 male on the unit) and the Terminal connector (db15 female) goes to Asterisk. These units started appearing on Ebay at the under $100 price mark about a year ago or so. Nice tip, though I won't be buying one for each of my 26 spans. :-) To the OP: if you're willing to open the box, there is usually a DB-9s on the front of the smartjack you can plug your laptop into... it's not actually your interface, but it will give you the numbers. You also run the risk of screwing up the span, which I do not assume hereby. :-) Cheers, -- jra Around here, the telco locks that cabinet and there is a user id/password required to use that craft interface. (I have a key, but then I worked for the telco for 23 years and have a few insider tools laying around) Lyle Giese ___ -- 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] Error Counters on PRI Circuit
Joe Pukepail wrote: Is there a way to see error counts on the T1 of a PRI? Hooked up to asterisk via a digium TE122. Looking for something to make sure I'm not getting any CRC, framing or other errors on the T1. Using asterisk 1.4.19 and zaptel 1.4.10 Go on ebay and buy an ADC Kentrox DataSmart 658 for less than $100 dollars. The 658 has an ethernet port for management and grabbing the stats of the T1 line itself. For an ISDN T1 PRI, set channels 1-23 to T1 Voice and Channel 24 to T1 Data, otherwise the B channel won't come up. You may need a DB15male and a DB15female to RJ45 adapters as not all of these units on Ebay come with them. The T1 from the phone company connects to the Network connector(db15 male on the unit) and the Terminal connector (db15 female) goes to Asterisk. These units started appearing on Ebay at the under $100 price mark about a year ago or so. If you buy new, I think they are around $1200 in the US. And if you are not sure how to configure them, ADC is quite accommodating and I have configured more than one and can assist. Lyle Giese LCR Computer Services, Inc. ___ -- 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] Need some input for Quad T1 and channel banks.
Doug Lytle wrote: Don Pobanz wrote: Doug Lytle wrote on Monday, March 31, 2008 5:40 PM This does not sound right. If it is 2 PRIs then it should be 46 channels I may have the terminology incorrect. I don't have a D channel, so I guess this would be called a T1 then? Doug A channel bank does not do ISDN. You will be using what is called a channelized T1. You will probably set it up as 24 voice channels useing ESF B8ZS. When you use a channelized T1, each channel carries it's own signaling state and called number info is sent over the voice path(unless you have rotatory phones). Caller ID is also sent out via the voice channel. ISDN puts all the signaling on a single data stream called the D channel and you need to have two phone switches that talk to each other over the D channel. The signaling channel carries the calling and called number as well as the busy/idle state for each of the voice channels. ___ -- 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] Telemarketer Torture....
James Finstrom wrote: Anyone have the telemarketer torture prompts? I would seriously like to revive this. Weasels and Monkeys work well for this. I put up one extension that uses Monkeysintro then Monkeys and loops. The other extension uses somethingwrong then weasels and again loops back around. I just forward them to one of those two extensions. If callerid worked more reliably I would automate it. But I get a lot of caller id failures on my incoming POTS lines, esp when calling in from my cell phone. Lyle ___ -- 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] Mail Server
Mike, Most newer Linux distro's use Postfix. It's simple to setup Postfix to use SMTP AUTH to send email. You need to figure out why the primary mail server is rejecting the emails and go from there. Contact me off list if you want more info. I think I have a quick how-to I wrote for myself on how to set Postfix to use SMTP AUTH when sending email. Lyle Giese LCR Computer Services, Inc. Mike Hammett wrote: I am the ISP. ;-) I'll have to look into that smarthost deal as there is no reverse DNS at this time (my upstream's server times out). -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Erik Anderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 13, 2008 4:25 PM Subject: Re: [asterisk-users] Mail Server On Thu, Mar 13, 2008 at 4:04 PM, Mike Hammett [EMAIL PROTECTED] wrote: I need to setup a small mail server on a local network. It only needs SMTP ability as it's just so Asterisk can send out emails. The machine has sendmail installed. My primary mail server seems to be rejecting the messages. Some research says something isn't configured properly. What do I have to do so the outside world accepts emails from my Asterisk box? It is behind a NAT. Does your ISP provide an SMTP server you can use? If so, it's usually easiest to set that up as a smarthost and tell sendmail to send through that server. If this isn't an option, you need to make sure that your asterisk server has a valid publicly-available DNS record (and reverse DNS). That's most likely the reason the remote server is rejecting these emails. -erik ___ -- 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: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue
If you take Asterisk down, the PRI should go down as the D channel is down. Then the telco should KNOW that there is trouble with the PRI and those channels are in trouble busy and not availible. If the telco still tries to push a call to a channel on a PRI that is down, then the telco is at fault. Lyle Matt wrote: That does sound like what is happening.. Telco knows channel 1-23 are not busy (so far as they are concerned), however.. so far as you are concerned, they are busy.. so telco sends the call down... but the equipment doesn't take it. I would *think* the Telco could keep trying channels down the hunt group, but maybe not? We have, in the past, seen this issue with our dial-up modem banks.. especially if I would take one offline. However, it is not a big enough issue (i.e. we don't take things down that often) for me to look into it fully. On Thu, Feb 14, 2008 at 4:07 PM, Don Kelly [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I think the problem is that the telco presents the call on a specific channel, then zaptel tells it that the channel is busy. We need to be able to tell the telco that each unused channel on a given span is unavailable, and it will determine that the others are in use and will present the call on a channel on another span. A rather ugly work-around (since Andrew seems to have lots of channels available, and one would assume that maintenance of this nature would occur during slow periods) would be to make calls to a DID in the same trunk group on all idle channels on the span shutting down then, when all channels on the span are in use and none of them are doing anything useful, take the span down hard so the telco will divert all calls to another span. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax *From:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] *On Behalf Of *Matt *Sent:* Thursday, February 14, 2008 2:28 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue Honestly.. this sounds like a telco issue.I understand what the other person is saying about the PRI still being technically up... BUT... if the channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to the next available channel, which they clearly are not doing. On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Tim, Imagine the scenario where we had 10x Asterisk servers, with calls presenting sequentially starting from the first server, then server two, etc. If we took down the first server for maintenance with 'asterisk -rx stop gracefully' we then will block all incoming calls to all servers as our telco will simply relay the BUSY back to the caller. If there are a number of calls on the first server that continue for another 20 minutes, then all inbounds are blocked for that period of time. We are finding at present we have to look at the calls on the server and make a decision if we are busy to simply reboot the server and hence lose calls. Not ideal but then we don't end up blocking our inbounds. What I was hoping to do was find a way to cause the telco to present the call to the next ISDN30 and therefore would allow us to cleanly take down an Asterisk server for maintenance without causing this issue. In a sense to put the ISDN30 into alarm mode while still continuing the active calls. Do you know if this is at all possible, even if we considered patching zaptel to add this functionality or does the telco rely on the entire PRI being in alarm before it presents the call to the next ISDN30 ? This would allow us to run maintenance on our servers during busy periods without causing disruption, and would be an excellent feature. Many thanks, Andrew *From:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] *On Behalf Of *Tim Nelson *Sent:* 13 February 2008 18:12 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue Even if * is shutdown, zaptel is still running and your ISDN
Re: [asterisk-users] IAX Calls - One Way Audio
Why not give the receptionist a two line phone? Register one line on server 1 and the other on server 2. Then the bounce back and forth goes away saving bandwidth. Lyle Daniel Cole wrote: Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for the whole organization, answering calls that come in for both locations. We have a problem where some calls (seemingly randomly) appear to get one way audio. This only happens for inbound calls off the PSTN, if they follow this pattern (which is a fair number of calls): Call comes in from PSTN to site A, gets put into a queue to be answered by receptionist as site B. Receptionist answers the call, and then puts the call on hold to perform an attended transfer to an extension at site A. (The call from the receptionist to the extension is OK). When the receptionist hits the 'transfer' button to actually transfer the call, the original caller cannot hear anything. The internal extension can hear the caller OK. This problem does not occur on every call. Since the issue has risen its head, I have enabled core, sip and iax debugging, but I am of yet unable to get the issue to occur on its own, to have a good look at the log files. FYI, I have disabled the asterisk Dial Commands in FreePBX, to solve another issue (where call audio bounces between the servers for a call that is transferred between sites and back again). Both servers are asterisk version 1.2.23, freepbx version 2.3.1.0. I have posted the contents of the iax.conf file below (which is identical on both servers). If there is any further information I can provide, please let me know and I can get this information. [general] disallow=all allow=g729 mailboxdetail=yes jitterbuffer=no ;maxjitterbuffer=500 ;jittershrinkrate=1 bandwidth=low tos=lowdelay trunk=yes notransfer=yes #include iax_general_custom.conf #include iax_registrations_custom.conf #include iax_registrations.conf #include iax_custom.conf #include iax_additional.conf Any suggestions are very welcome. Regards, Daniel ___ -- 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] Finding difficulty in installing Asterisk
You need to do a 'make' before the 'make install'. Lyle [EMAIL PROTECTED] wrote: Hi all, Please help me in installing Asterisk. I am getting the following error when trying to install Libpri [EMAIL PROTECTED] Asterisk]$ cd libpri-1.4.2 [EMAIL PROTECTED] libpri-1.4.2]$ make clean rm -f *.o *.so *.lo *.so.1 *.so.1.0 rm -f testprilib libpri.a libpri.so.1.0 rm -f pritest pridump rm -f .depend [EMAIL PROTECTED] libpri-1.4.2]$ make install gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -c -o copy_string.o copy_string.c ___ -- 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] no outgoing calls with Digium B410P
daniele visaggio wrote: 2008/1/7, map [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi Daniele, Please send a snapshot of your Putty Asterisk log. Go to Putty configuration - Window - Lines of scrollback and put a number greater than 200 :-). I suggest 10. Sorry, i'm using a Linux version of putty (i'm running Ubuntu) and the configuration is different. I can't find Lines of scrollback and modify the scrollback number. The putty-linux sw-structure is probably different from the putty-windows one. ___ --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 Since you are using linux, open an Xterm window and issue the following command: ssh -l user id name or ip address of * This should prompt you to verify the ssh key the first time and then ask for your password. Cut and paste works from an XTerm window. Lyle ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
Olle E Johansson wrote: All I can say is with 1.6, if a change is made that causes something that worked in 1.4 not to work in 1.6, please think twice, three times or four times before making the change, or making the change in such a way that it won't break dialplan stuff from 1.4. Our policy is to never remove any functionality between two versions. We replace the functionality with new functionality and print out warnings whenever you use the deprecated functions. We also add this to the documenation in the software and the UPGRADE.TXT file. So the functionality that you lost in 1.4 was old 1.0 functions that was marked as deprecated in 1.2 and removed in 1.4. We might want to be more informative about those changes. We need to make a clear list of things you need to start changing as a user of 1.4 to prepare for lost functionality in 1.6. This information already exist, but should maybe be a bit more public. In some cases we do have to change in a dramatic way and can't preserve the old functionality to solve a bug in the software. This requires thorough discussion in the developer group and is something we really want to avoid at all costs. If this happens, it's clearly documented in the software. Thank you for your feedback, it's important to us. /O Along that this same line, I ran 1.0.something for a long time and it was working just fine for my SOHO. I had a channel bank to interface pots lines from the local Telco and feed the analog phones in the house. Over time, I replaced most of those analog phones with SIP phones. An unfortunate incident caused us to lose that server and several sip phones. When I recovered enough to rebuild *, I tried 1.4 and it would not compile completely and zaptel did not load properly. I download 1.2 and it worked with the same configs as 1.0, but the quality was poor. That was due to hardware issues. I purchased a new motherboard and rebuilt using a newer Asterisk 1.4 with the then current libpri and zaptel and the call quality came back. But I had a hard time with syntax changes. Basically I was jumping from 1.0.x to 1.4.x in one leap. My biggest gripe is that everything loaded and seemed to work. A day later we found this did not work and discovered a syntax change. A day later something else did not work, an other syntax change. Why isn't there some pre-processor to check the syntax of the config files? Would have saved me a whole bunch of time I didn't have to spare and still don't. Lyle As it is syntax problems or changes are not noticed or logged until Asterisk tries to execute them. If there is a chunk of code that is only hit once a week??? It almost came to a point of scraping Asterisk because of the push back from the family. ___ --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] Using existing extensions.conf macros, and co-habitation
Brian J. Murrell wrote: On Fri, 2007-11-30 at 15:08 -0800, Philip Prindeville wrote: bump... What's with all this bump I see here? Is this a web forum? b. ___ --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 Somebody asked a question and no one answered. A bump is just a nudge to politely ask this is the 2nd time I have asked this does someone know the answer. I have used the before and it usually works. Lyle ___ --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] Voice mail Uniden UIP-200 phones
Yep, that fixed it. Just shaking my head as to why the behavior changed... Lyle CunningPike wrote: Try dtmfmode=inband CP Lyle Giese wrote: I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect' from Allison. I go to any other phone and I can log in just fine. Just not from our two Uniden phones. I have no problem placing calls. In the messages log, I see: app_voicemail.c: Unable to read password or app_voicemail.c:Couldn't read username Again, going to a different phone other than one of my two Uniden phones and no problem accessing and retreiving voicemail. In sip.conf against the UIP-200's I have: nat=never dtmfmode=rfc2833 Otherwise, I stayed with the standard Uniden provided config files served up via tftp and only made the minimum required changes to config files in Asterisk. I am running firmware 4.77(also tried downgrading firmware on phones to 4.63). Any suggestions? Thanks, Lyle Giese ___ --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: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice mail Uniden UIP-200 phones
I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect' from Allison. I go to any other phone and I can log in just fine. Just not from our two Uniden phones. I have no problem placing calls. In the messages log, I see: app_voicemail.c: Unable to read password or app_voicemail.c:Couldn't read username Again, going to a different phone other than one of my two Uniden phones and no problem accessing and retreiving voicemail. In sip.conf against the UIP-200's I have: nat=never dtmfmode=rfc2833 Otherwise, I stayed with the standard Uniden provided config files served up via tftp and only made the minimum required changes to config files in Asterisk. I am running firmware 4.77(also tried downgrading firmware on phones to 4.63). Any suggestions? Thanks, Lyle Giese ___ --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] Help: Asterisk info
And why are you asking in the Asterisk list? The absence of that file means you don't have any scsi adapters in your system. Lyle Jarga Jallow wrote: I am getting this error under system info: File Line Command Message common_functions.php 314 file_exists(/proc/scsi/scsi) the file does not exist on your machine Does anybody know how to fix this? Thank you in advance Jarga ___ --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
[asterisk-users] [Fwd: voicemail locked up Asterisk 1.4.13]
The orginal did not make it to the list... Spam filter issue??? No repeat of the lockup yet. Lyle Original Message Subject:voicemail locked up Asterisk 1.4.13 Date: Thu, 01 Nov 2007 20:57:27 -0500 From: Lyle Giese [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com I am running Asterisk 1.4.13 with libpri 1.4.2 and zaptel 1.4.6 on openSuSE 10.2 (64bit kernel) with an AMD dual core 64 bit processor at 2ghz and 1g of ram. Motherboard has a VIA chipset. Using an Adtran chan bank to interface the incoming POTS lines via a Digium single T1 pci card. Very lightly loaded system used in a SOHO environment. Asterisk ran flawlessly under 1.0.x for a long time on an older motherboard with this chan bank and T1 card. Today, we started getting callerid failures on incoming calls(checksum errors and invalid callerid errors). On one line, I had programmed * to send invalid or missing CallerID directly to voicemail without ringing any phones. We went out for lunch and came back and found 4 new voice messages. However voicemail would not let us log in. We called voicemail, inputed the vm box number and were prompted for the password and punched it in. And all Allison would say is 'login incorrect'. All voicemail boxes. We were locked out of voice mail. No error messages anywhere that I can find. Restarted Asterisk and VoiceMail now works. Logged in and listened to the voice messages, etc. I know, it's not much to go on, but is there something I can set to get more verbose error messages, if this happens again? Thanks, Lyle Giese LCR Computer Services, Inc. ___ --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] Uniden UIP200 phones
Mojo with Horan Company, LLC wrote: Lyle Giese wrote: Philipp Kempgen wrote: Lyle Giese wrote: I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2) SIPAddHeader(Alert-Info: ...); Regards, Philipp Kempgen Took me a while to notice the difference between - and _ But it works now! Do you mean you're using SetVar(Alert-Info: ...) instead of SIPAddHeader(Alert-Info: ...) ? Thanks, Moj I WAS using SetVar with * v1.0.x. For version 1.4.x, I had to ask what the new syntax was for the same functionality. Lyle ___ --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] Uniden UIP200 phones
Philipp Kempgen wrote: Lyle Giese wrote: I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2) SIPAddHeader(Alert-Info: ...); Regards, Philipp Kempgen Took me a while to notice the difference between - and _ But it works now! Thanks, Lyle ___ --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] Treating T1 as trunk in/out, not individual lines
The same as any other zap channel does. That is part of the magic of the zaptel drivers. Lyle Michelle Dupuis wrote: Ok..so how would the CALLED and CALLERID ID be presented to Asterisk when using PRI signaling. Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Lyle Giese *Sent:* Friday, October 26, 2007 5:54 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Treating T1 as trunk in/out, not individual lines Michelle Dupuis wrote: I'm tying a Nortel option 61 to asterisk via T1. I don't want to split each of the t1 channels out into individual lines (tied to a specific extension) - so a trunk in and out. Assuming PRI over T1 signaling, how would I pass the CALLED and CALLER info across the channels so each side knows what to do? Is there something in the PRI protocol you can point me to for figuring this out? Thanks, MD ___ --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 That's what the D channel is for. A PRI is a primary rate ISDN. B channels carry voice, D channel handles the information signalling in ISDN. Lyle ___ --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
[asterisk-users] Uniden UIP200 phones
I am trying to get distinctive ringing going again with these phones, depending on the outside line the call comes in on. I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2) But it doesn't give me the ring I want on the phone. I have firmware BS4.63 and BS4.77 on the phones and it doesn't seem to work on either. Any suggestions? Thanks, Lyle ___ --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] ABE, Sangoma, T-1 no recognizing calls
Your signalling is wrong. The channels as programming in * should fxsks (use ks instead of ls) and not fxols. At Verizon's end, they use fxo and you grab it via fxs emulation in *. Lyle John Millican wrote: Hello All, I have a setup of ABE on rPath linux,Sangoma A101D, and a T-1 line (Not PRI) which is all happily coexisting and all lights are green. The T-1 comes in from the world into a Shark Box which splits the T into 384K data and 6 channels voice. The data side is working great. The voice side, not so great. It was originally broken out to 6 pots line and Verizon came back and swapped cards in the shark and now it is a T-1 out. Wanrouter, zaptel and asterisk are all apparently happy. When I place a call to * I hear ring on the calling side but do not ever see anything in happen on the * side. When I try to call out i get: Executing Dial(SIP/xxx.xxx.xxx.xxx-ab5012d0, zap/3/603xxx) in new stack -- Called 3/603xxx And nothing else, at one time I was getting a zap/answered line but no more. Relevent zapata.conf [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A101 port 1 [slot:8 bus:3 span: 1] context=from-pstn group=0 signalling=fxo_ls channel = 1-6 zaptel.conf loadzone=us defaultzone=us #Sangoma A101 port 1 [slot:8 bus:3 span: 1] span=1,1,0,esf,b8zs fxols=1-6 Extensions.conf [from-pstn] exten = _X.,1,Dial(zap/3/603xxx); Very simple setup at this moment, nothing fancy. I am able to dial in via sip and asterisk answers and send the call to the from-pstn context at which point i see Executing Dial(blah, blah) in new stack; I believe at this time that the problem is in the setup of the shark box. Verizon tells me that there end is good and the T-1 is esf, B8ZS, loop start. But I thought I would ask the list for some opinions before I started pointing the finger. Thank you for any help JohnM ___ --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] Need T1 crossover cable?
Michelle Dupuis wrote: I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My Sangoma A102D shipped with 2 T1 cables - which I assume are straight through. Do I need to make crossover cables for this scenario? Thanks ___ --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 Yes, use a T1 crossover(not an ethernet crossover). Lyle ___ --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] Treating T1 as trunk in/out, not individual lines
Michelle Dupuis wrote: I'm tying a Nortel option 61 to asterisk via T1. I don't want to split each of the t1 channels out into individual lines (tied to a specific extension) - so a trunk in and out. Assuming PRI over T1 signaling, how would I pass the CALLED and CALLER info across the channels so each side knows what to do? Is there something in the PRI protocol you can point me to for figuring this out? Thanks, MD ___ --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 That's what the D channel is for. A PRI is a primary rate ISDN. B channels carry voice, D channel handles the information signalling in ISDN. Lyle ___ --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] Need to run ztcfg manually?
Zaptel creates a startup script. You just need to make sure it run/loads fully before Asterisk starts in your bootup scripts. This gets into tweeking your system and that varies based on the exact OS/distro you are running. Lyle Mojo with Horan Company, LLC wrote: I don't have T1 but it seems that the first time I run ztcfg (or in fact, the zaptel startup script runs it for me) it fails. Then I need to run it again for it to actually configure things right. So, my (redhat-style) /etc/rc.d/rc.local contains modprobe wctdm ztcfg -vv asterisk Michelle Dupuis wrote: I have a new asterisk system with a T1 card. It appears that running ztcfg -vv is required in order for asterisk to start properly. Is this correct? Are people adding this command to the asterisk startup script? Thanks ___ --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: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E4 Superframe EM?
Steve Totaro wrote: Richard Lyman wrote: Steve Totaro wrote: I need to create a couple of tie lines between a legacy system and an Asterisk system. I was told that the tie lines are E4 Superframe EM. I have done EM wink but have no idea about E4 Superframe EM and Google is not helping me here. Does anyone know about this type of signaling and if Asterisk can handle it? Thanks, Steve use this zaptel.conf span=1,1,0,d4,ami em=1-24 ; 1-32 for E4 zapata.conf signalling=em channel = 1-24 ; 1-32 for E4 Thanks to everyone who has responded so quickly to my question. To my way of thinking, it would be better to have the legacy tie-line reconfigured to use esf if possible. Is D4 (superframe) well supported in Asterisk, are there less features? If it is virtually the same, then I guess I will just setup Asterisk to use it rather than messing with the legacy system. Thanks, Steve ___ --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 D4 superframe - more bits are used for chan signalling, results in 56k voice channels. Extended superframe - fewer bits used for chan signalling, results in 64k voice channels. Otherwise basically the same, both are 1.544 mbps. If the T1 goes down, D4 takes longer to frame up than ESF. But for pbx to pbx, aren't they going to be located in the same room? Lyle ___ --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
