Re: [Asterisk-Users] PABX and Asterisk Dial Plan
You will want to use the D(digitstopluseindtmf) option on your dial cmd. That is a capital D for the option! ex. Dial(SIP/2100,D(1000)) -Jon Stephen wrote: Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone - Asterisk -- ATA (FXS) -- (CO side) PABX - Extension (eg. 1000) (2100 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by dialing 2100 from sip phone after PABX answer my call. But that's too troublesome, Can I just dial 21001000 instead ? which mean the first 4 numbers are for pabx and the next 4 numbers are for extension? Thanks, Stephen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk Installations + SER
On Mon, Aug 15, 2005 at 07:54:50AM +0200, Ronald Voermans wrote: Okay, First of all, thank you for your input. I didn't know that I could use 1 * for multiple companies (wish I knew it earlier, because installing vserver and installing * on a vserver took me a lot of time :) ). Nevertheless, I think I still will need the SER. If my 'shared *' server is getting overloaded, I want to be able to quickly add a new * server. For the IP Voice Interconnect to work properly, I think I need one 'gateway' on our side, which will be SER. Is this correct? Those asterisk instances still share quite a few resources: the network bandwidth and probably the CPU time. With some scriptology, it would probably be rather simple to add another company to your Asterisk configuration. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sirrix bri card:killing the machine
Hi all, I have a sirrix bri card, connected to 2 ISDN lines.I used to get a lot of slips on it, so i put the 'master' setting to 'yes'. But every couple of hours the machine completly hangs, and i have to reboot it. I only get 1 or 2 slips,I dont know whats wrong? yusuf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple Asterisk Installations + SER
If I install 1 * server, with multiple companies/dialplans, how do I make 1 company dial the other company with a full telephonenumber (i.e. 10 digits)? -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Tzafrir Cohen Verzonden: maandag 15 augustus 2005 8:28 Aan: asterisk-users@lists.digium.com Onderwerp: Re: [Asterisk-Users] Multiple Asterisk Installations + SER On Mon, Aug 15, 2005 at 07:54:50AM +0200, Ronald Voermans wrote: Okay, First of all, thank you for your input. I didn't know that I could use 1 * for multiple companies (wish I knew it earlier, because installing vserver and installing * on a vserver took me a lot of time :) ). Nevertheless, I think I still will need the SER. If my 'shared *' server is getting overloaded, I want to be able to quickly add a new * server. For the IP Voice Interconnect to work properly, I think I need one 'gateway' on our side, which will be SER. Is this correct? Those asterisk instances still share quite a few resources: the network bandwidth and probably the CPU time. With some scriptology, it would probably be rather simple to add another company to your Asterisk configuration. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to load module for TE406P
Hi, I'm unable to load wct4xxp module for TE406P card. I've compiled 2.6.13-rc6, got latest CVS (as of today) of zaptel, but when I try to load the module I get this: kobject_register failed for Unified t4xxp/t2xxp driver (-13) [kobject_register+53/73] kobject_register+0x35/0x49 [bus_add_driver+62/153] bus_add_driver+0x3e/0x99 [driver_register+55/58] driver_register+0x37/0x3a [pci_register_driver+120/134] pci_register_driver+0x78/0x86 [pg0+945848335/1069265920] t4_init+0xf/0x22 [wct4xxp] [sys_init_module+199/462] sys_init_module+0xc7/0x1ce [syscall_call+7/11] syscall_call+0x7/0xb Can anyone point me into right direction to solve this? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting 2 * servers
Hi List Does anyone have the configs to connect 2 * servers so that clients from the one * server can make pstn (ZAP) calls out from the other * server ? Thanks in advance Sean ___ Join Excite! - http://www.excite.com The most personalized portal on the Web! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with FWD connection rejected
Sean Rima wrote: Using the install instructions for [EMAIL PROTECTED], I setup a FWD account, this I tested using X-Lite and it works okay, Nowever I cannot make calls to fwd using Asterisk, my log showes: Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected: Registration Refused Aug 14 21:06:59 NOTICE[1324]: Registration of '689482' rejected: Registration Refused I'm new to this, but just a couple of thoughts: were you using SIP to connect to FWD from XTEN? Are you using SIP or IAX to connect to FWD from Asterisk? If you're using IAX, have you done the additional registration step needed by FWD to enable the IAX protocol? It might be worthwhile posting your configuration (sip.conf or iax.conf depending on what you're using). John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO port trhoug optimum voice VOIP service
I have 2 OEM X100P. The one from www.broad-tel.com works fine.However, the other one has echo. Both use MD3200 chips. Any one knows why it is so?? On 8/13/05, Madhawa Jayanath [EMAIL PROTECTED] wrote: Carlos Trallero wrote: Hello, I have asterisk running on Fedora Core 3 with a x100p (oem). After some time I got asterisk with some soft extensions working (u gotta love open source), but I'm stuck with outbound dialing. This is the diagnose: - detect 1 wcfxo channel. - when trying to make an outside call I get unable to create channel of type Zap. Everyone is busy/congested at this time - When I plug the x100p to the phone jack, the dial tone in all of my phones die. Because of the later I'm suspecting that there is some problem with the signaling or voltage detection. My PSTN line is actually from a VoIP modem that runs over the Cablevision network (known as Optimum Voice). Thanks everyone. Carlos __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, Where did u get that OEM X100P? Is it MD3200 chip? Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] permission denied when monitoring channel OSS/dsp
Hi! When I want to monitor the OSS/dsp channel through the Asterisk management interface, I get a permission denied error: Action: Monitor Monitor: OSS/dsp File: 1124096949 Mix: 1 Response: Error Message: Permission denied My permissions for /var/spool/asterisk look like this: pound:~# ls -la /var/spool/asterisk/ total 40 drwxr-xr-x 10 asterisk asterisk 4096 Aug 9 10:19 . drwxr-xr-x 9 root root 4096 Aug 4 14:40 .. drwxr-xr-x 2 root root 4096 Jul 8 09:40 dictate drwxr-xr-x 2 root root 4096 Jul 8 09:40 meetme drwxr-x--- 2 asterisk asterisk 4096 Aug 9 10:32 monitor drwx-- 2 root root 4096 Jul 4 16:56 outgoing drwxr- 2 root root 4096 Jul 4 16:56 qcall drwxr-xr-x 2 root root 4096 Jul 8 09:40 system drwxr-xr-x 2 asterisk asterisk 4096 Mar 21 12:23 tmp lrwxrwxrwx 1 root root 37 Jul 13 10:08 vm - /var/spool/asterisk/voicemail/default drwxr-xr-x 3 asterisk asterisk 4096 Jul 4 16:53 voicemail So, there shouldn't be any problems writing to disk. Anything else that I need to take into account, especially something special about monitoring the OSS/dsp channel? Thanks, Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco and protocol application invalid
On 8/14/05, Bjorn Ove Kristiansen [EMAIL PROTECTED] wrote: Is there any way of getting to know which IP address Cisco uses to contact TFTP? Why you're making things hard for yourself for no good reason? Unlock the config, put a static entry for ip that belongs to the segment its sitting on (RFC1918 or not), put a static entry for your TFTP server and upgrade. When all goes well setup your DHCP pools with the tftp options for future use. Regards, Bjorn Hope that helps a bit nkm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Security and SIP
I've now setup SIP for: - internal softphones - registering with external providers (like FWD) for making calls - receiving calls from theese providers For the latter step, it was necessary to forward ports from my NAT to the asterisk server: 5060 + range of ports mentioned in rtp.conf. I was just wondering about how to make this setup as secure as possible. Here's what I've done so far: 1. defined a default context in sip.conf which cannot access any real extension. sip.conf: [general] context=from-unknown-sip extensions.conf: [from-unknown-sip] exten = _.,1,CONGESTION 2. for peers, defined a context which does not provide access to outside lines. sip.conf: [fwd.pulver.com] type=peer username=688426 fromuser=688426 secret=xx host=fwd.pulver.com port=5060 nat=yes canreinvite=no insecure=very context=sip-external disallow=all allow=ulaw 3. for peers, defined insecure=very which should check that the incoming call comes from the same IP as was registered. 4. for internal softphones, which can make outgoing calls, limited registrations to a specific network address using deny/permit sip.conf: [31] type=friend callerid=[EMAIL PROTECTED] 31 host=dynamic deny=0.0.0.0/0.0.0.0 permit=192.168.2.32/255.255.255.255 context=sip-internal secret= disallow=all allow=ulaw allow=alaw Anything else I can do to improve security? I specifically don't want anyone external to be able to make calls. As I've opened port 5060 + rtp.conf ports only for the purpose of receiving calls from services I have registered with, I don't want any external phones to be able to register via this route. Is there any risk of this if someone can guess a password (maybe unlikely but given time this could happen). Thanks, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with FWD connection rejected
John Fawcett wrote: Sean Rima wrote: Using the install instructions for [EMAIL PROTECTED], I setup a FWD account, this I tested using X-Lite and it works okay, Nowever I cannot make calls to fwd using Asterisk, my log showes: Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected: Registration Refused Aug 14 21:06:59 NOTICE[1324]: Registration of '689482' rejected: Registration Refused I'm new to this, but just a couple of thoughts: were you using SIP to connect to FWD from XTEN? Are you using SIP or IAX to connect to FWD from Asterisk? If you're using IAX, have you done the additional registration step needed by FWD to enable the IAX protocol? It might be worthwhile posting your configuration (sip.conf or iax.conf depending on what you're using). I have just scratched the setup and will be using the setup that is on the wiki to try it out, the only problem is that I am dialup and cannot use the externalip= setting, so have to work a way around that as yet Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Java-Call Problem
Hello,
i have some problem with the Asterisk and need a little bit help.
I have an dialplan like this:
[first]
.
..
...
exten = s,6,Read(Secret,,25)
exten = s,7,NoOp(**${Secret}**)
exten = s,8,Gotoif($[${Secret} = 999]?9:9)
exten = s,8,Wait(15)
When the user enter the secret key, my java- backend application validate the
entered user input.
If the user input is correct, i want that the user comes to the second context
controlled by my java application:
[second]
.
..
...
But how i can i send the next context information to an open phone line ?
I can't find any Java examples on the net.
If you have any idea / examples?
Thanks
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Re: [Asterisk-Users] Multiple Asterisk Installations + SER
Hi, Ronald Voermans wrote: If I install 1 * server, with multiple companies/dialplans, how do I make 1 company dial the other company with a full telephonenumber (i.e. 10 digits)? This is very much dependant on how your dialplan works. We use normalisation for each account so the system doesn't have to worry about many different dialling formats (i.e. with or without areacode, and such). You can use a similar strategy for all your internal numbers as well. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PC for 8 line system
Chris Gamble schrieb: I have 2 TDM04b cards currently running in an asterisk at home box that I am ready to replace with the CVS version of asterisk. What I am looking for is thoughts / recommendations. I want to move this to a small form factor ( shuttle ) machine and was wandering what expeience / advice there was for this? I have seen the incompatible motherboard list at digium ( and in fact I think my current machine is on the list ! ), but wanted to know what others are doing for small form factor tdm setups? I really love the Asus Pundit-R. It has on board VGA and 2 PCI-Slots. Installing Linux on it is a bit tricky but if you don't need the builtin cardreader Debian Sarge works like a charm on it. We use them in production enviroment (as office PBXes) with no problems. Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Voice mail to nortel PBX Option 11
Of course you do. How would the * system know whom the call is for? Mark craz sead wrote: Do i have to create voice mail one by one (per ext at the meridian) at the * box ? --- Mark Phillips [EMAIL PROTECTED] wrote: Easily doable. I've done it twice now. Problem is that your users will never know they have messages waiting. Install a T1/E1 card into the * box and then use a T1 cross-over cable between the 2 boxes. Create a dialplan on the Meridian that points calls to the VM out over the new E1. As for forwarding the calls when busy or no answer, that's a little more tricky. You'll have to come up with some rules and numbers to allow the Meridian to decide what to do with those calls. In my case I wrote a forward on no answer and a forward on busy rule for every phone that needed VM. When you called ext 200 the call was sent to mailbox 2200 on the *. Users will have to get into the habit of calling the VM to check if there's messages. Hope that helps. Mark craz sead wrote: Hi all, Could somebody help me, i wanna connect asterisk for voice mail in the existing nortel pbx option 11 using e1 card ? anyone have a clue ? please help the conf. file thank all __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TELASIP DOWN?
Got the same thing, thanks Jeff. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, August 14, 2005 12:00 PM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 13, Issue 98 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com Its back up! According to Support Sorry for the inconvenience, Since Friday we have had an issue with someone flooding our servers with requests. This resulted in failed registrations and failed call completion. Our service is available again, but we are continuing to work the issue. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] permission denied when monitoring channel OSS/dsp
On Monday 15 August 2005 11:11, Christoph Eicke wrote: Hi! When I want to monitor the OSS/dsp channel through the Asterisk management interface, I get a permission denied error: Action: Monitor Monitor: OSS/dsp File: 1124096949 Mix: 1 Response: Error Message: Permission denied it's always nice to answer your own posts ;-) The Permission denied message had nothing to do with file attributes, but with what is written inside of the /etc/asterisk/manager.conf users's section that is connecting to the Asterisk management interface. There you have to set the right permissions. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting 2 * servers
All security issues aside here, this is very easy. Create an IAX trunk between the 2 servers. Put the Zap side trunk into the same context that is allowed to dial the Zap line. On the non-Zap server create a dialplan rule that forwards all calls to the PSTN over the the Zap host. Why do we want a IAX trunk rather than a SIP trunk? To save on some bandwidth. Hope this helps. If not write me off list. Mark Sean wrote: Hi List Does anyone have the configs to connect 2 * servers so that clients from the one * server can make pstn (ZAP) calls out from the other * server ? Thanks in advance Sean ___ Join Excite! - http://www.excite.com The most personalized portal on the Web! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO port trhoug optimum voice VOIP service
Are these cards on the same line or are they on different lines. Assuming different lines, what happens when you swap the lines over? Does the echo follow the line? Also, It seems that these cards can be modified to do a number of things either in firmware or in hardware. Are they electronicly identical. If so then are they the same firmware version? I'm just throwing them out there. I have the same problem and my cards identify themselves as different manufacturers. I'm assuming that they have different firmware versions from the OEM. Mark VoIP Newbie wrote: I have 2 OEM X100P. The one from www.broad-tel.com works fine.However, the other one has echo. Both use MD3200 chips. Any one knows why it is so?? On 8/13/05, Madhawa Jayanath [EMAIL PROTECTED] wrote: Carlos Trallero wrote: Hello, I have asterisk running on Fedora Core 3 with a x100p (oem). After some time I got asterisk with some soft extensions working (u gotta love open source), but I'm stuck with outbound dialing. This is the diagnose: - detect 1 wcfxo channel. - when trying to make an outside call I get unable to create channel of type Zap. Everyone is busy/congested at this time - When I plug the x100p to the phone jack, the dial tone in all of my phones die. Because of the later I'm suspecting that there is some problem with the signaling or voltage detection. My PSTN line is actually from a VoIP modem that runs over the Cablevision network (known as Optimum Voice). Thanks everyone. Carlos __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, Where did u get that OEM X100P? Is it MD3200 chip? Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Security and SIP
You could make your FWD sonfigs even more secure by switching to IAX (you have to register with them for it) and then you can use RSA keys (already in your * distro) to prevent faking of connections. Check with the FWD site. Ther's a howto on there. I use this method and I like it alot. Especially as the IAX trunk allows me to have more than one concurrent call and takes up very little extra network overhead. Mark John Fawcett wrote: I've now setup SIP for: - internal softphones - registering with external providers (like FWD) for making calls - receiving calls from theese providers For the latter step, it was necessary to forward ports from my NAT to the asterisk server: 5060 + range of ports mentioned in rtp.conf. I was just wondering about how to make this setup as secure as possible. Here's what I've done so far: 1. defined a default context in sip.conf which cannot access any real extension. sip.conf: [general] context=from-unknown-sip extensions.conf: [from-unknown-sip] exten = _.,1,CONGESTION 2. for peers, defined a context which does not provide access to outside lines. sip.conf: [fwd.pulver.com] type=peer username=688426 fromuser=688426 secret=xx host=fwd.pulver.com port=5060 nat=yes canreinvite=no insecure=very context=sip-external disallow=all allow=ulaw 3. for peers, defined insecure=very which should check that the incoming call comes from the same IP as was registered. 4. for internal softphones, which can make outgoing calls, limited registrations to a specific network address using deny/permit sip.conf: [31] type=friend callerid=[EMAIL PROTECTED] 31 host=dynamic deny=0.0.0.0/0.0.0.0 permit=192.168.2.32/255.255.255.255 context=sip-internal secret= disallow=all allow=ulaw allow=alaw Anything else I can do to improve security? I specifically don't want anyone external to be able to make calls. As I've opened port 5060 + rtp.conf ports only for the purpose of receiving calls from services I have registered with, I don't want any external phones to be able to register via this route. Is there any risk of this if someone can guess a password (maybe unlikely but given time this could happen). Thanks, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO port trhoug optimum voice VOIP service
To make sure no configuration issue, I only had one of them working at a time. After test on one of them, I swapped out with the other and rebooted. Both were detected as same. However, echo only happened on one but not the other. I know nothing about electronics. Layout of both cards looks almost the same. However, the card from www.broad-tel.com never have echo during several tests. the one I got from somewhere else has echo issue. I bought the second one from www.broad-tel.com since they claimed their cards never have experienced echo problem. And it does in my case. How can I tell their firmware version if any? On 8/15/05, Mark Phillips [EMAIL PROTECTED] wrote: Are these cards on the same line or are they on different lines. Assuming different lines, what happens when you swap the lines over? Does the echo follow the line? Also, It seems that these cards can be modified to do a number of things either in firmware or in hardware. Are they electronicly identical. If so then are they the same firmware version? I'm just throwing them out there. I have the same problem and my cards identify themselves as different manufacturers. I'm assuming that they have different firmware versions from the OEM. Mark VoIP Newbie wrote: I have 2 OEM X100P. The one from www.broad-tel.com works fine.However, the other one has echo. Both use MD3200 chips. Any one knows why it is so?? On 8/13/05, Madhawa Jayanath [EMAIL PROTECTED] wrote: Carlos Trallero wrote: Hello, I have asterisk running on Fedora Core 3 with a x100p (oem). After some time I got asterisk with some soft extensions working (u gotta love open source), but I'm stuck with outbound dialing. This is the diagnose: - detect 1 wcfxo channel. - when trying to make an outside call I get unable to create channel of type Zap. Everyone is busy/congested at this time - When I plug the x100p to the phone jack, the dial tone in all of my phones die. Because of the later I'm suspecting that there is some problem with the signaling or voltage detection. My PSTN line is actually from a VoIP modem that runs over the Cablevision network (known as Optimum Voice). Thanks everyone. Carlos __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, Where did u get that OEM X100P? Is it MD3200 chip? Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P / TE410P with 2nd generation firmware field upgradable?
Here's their press release with the improvements for firmware v2: http://www.digium.com/index.php?menu=press/pr_2gen_firm MATT--- On 8/15/05, Adam Goryachev [EMAIL PROTECTED] wrote: On Fri, 2005-08-12 at 08:29 -0500, Kevin P. Fleming wrote: Matt Florell wrote: We do 2nd gen firmware upgrades for customers every day, have been doing them for a couple of weeks now. The process is simple, just contact the RMA department and be prepared to pay for shipping the card to and from our facility. Once the card has 2nd gen firmware on it, future upgrades will be possible in place, without having to send the card to us. Apart from that (future firmware being software upgrade-able), what other reasons are there for people wanting to upgrade? ie, is there a changelog somewhere? Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 8304 0001www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP301 and 501 with asterisk...
We've deployed IP 300's, 500's, and 501's at customers and they work very well. On Thu, Aug 11, 2005 at 11:52:35AM -0700, Ing. Marlo R. Beltran G wrote: I am about to buy ip pbx asterisk system but what ip phones do you recommend? Are polycom ip all functional with the ip pbx system??? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk + chan_mISDN = undefined symbol: ast_pickup_call
Hi all! I'm getting an error when I try to start asterisk with chan_misdn. I patched my kernel (2.6.12.4), and compiled the whole stuff (kernel, mISDNuser, asterisk, chan_misdn). I got mISDN from http://isdn.jolly.de/download/v3.0/ I'm using a CVS Snapshot of asterisk, which was checked out about 5 hours ago. This is the error: [chan_misdn.so]Aug 15 14:13:29 WARNING[4929]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/chan_misdn.so: undefined symbol: ast_pickup_call Aug 15 14:13:29 WARNING[4929]: loader.c:488 load_modules: Loading module chan_misdn.so failed! I have no idea where to start solving this problem. Has anybody a hint for me? If you need more information, feel free to ask for it. Greets, Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple Asterisk Installations + SER
I'm not sure I understand what you mean... I want to have internal extensions (100, 101, 102, etc.) and some full phone-numbers (10 digits). How do I implement this in *? Ronald -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Florian Overkamp Verzonden: maandag 15 augustus 2005 11:58 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Multiple Asterisk Installations + SER Hi, Ronald Voermans wrote: If I install 1 * server, with multiple companies/dialplans, how do I make 1 company dial the other company with a full telephonenumber (i.e. 10 digits)? This is very much dependant on how your dialplan works. We use normalisation for each account so the system doesn't have to worry about many different dialling formats (i.e. with or without areacode, and such). You can use a similar strategy for all your internal numbers as well. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO port trhoug optimum voice VOIP service
Based on research that I did some time ago, there are multiple versions of the MD3200 chipset. One targeted for use in US telephone systems, and another targeted for non-US systems (that have different impedence matching requirements). Sounds like you have one of each. I have 2 OEM X100P. The one from www.broad-tel.com works fine.However, the other one has echo. Both use MD3200 chips. Any one knows why it is so?? On 8/13/05, Madhawa Jayanath [EMAIL PROTECTED] wrote: Carlos Trallero wrote: Hello, I have asterisk running on Fedora Core 3 with a x100p (oem). After some time I got asterisk with some soft extensions working (u gotta love open source), but I'm stuck with outbound dialing. This is the diagnose: - detect 1 wcfxo channel. - when trying to make an outside call I get unable to create channel of type Zap. Everyone is busy/congested at this time - When I plug the x100p to the phone jack, the dial tone in all of my phones die. Because of the later I'm suspecting that there is some problem with the signaling or voltage detection. My PSTN line is actually from a VoIP modem that runs over the Cablevision network (known as Optimum Voice). Thanks everyone. Carlos __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, Where did u get that OEM X100P? Is it MD3200 chip? Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codecs order
hi, i have this topology pstn+(e1)asterisk1-asterisk2-sip client asterisk1,asterisk2 allow (g729,alaw) sip client prefer g729, then alaw can you someone describe codec negotiation when call for sip client arrive from pstn? (can i set g729 for calls from pstn? ) thanks --- Marek Cervenka === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk Installations + SER
Brian Capouch wrote: I'm trying to figure out how to do some things like round-robin server balancing and the like using Realtime, and it seems like the right way to do it would be either via pre-processing the SQL requests coming in, or using stored procedures in the database that would accomplish the same thing. Not quite sure I understand the need to pre-process SQL requests. We just run 1 MySQL server on a RAID5 array. 1 webserver running the scripts which allow the customers to login and do their configuring. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
I am using the correct version of pwlib(1.5.2) and openh323(1.12.2) for the stable asterisk build. Both packages configure and compile with no problems. However when compiling chan_h323 from the asterisksource/channels/h323 directory I get this error. Chan-h323.h:31: warning; sockaddr_in bindaddr defined but not used arc r libchanh323.a ast_h323.o There have been posts of similar messages but none with the specific syntax as the one above. Those posts have mentioned commenting some part of the source code in order to build. I have combed the Makefile and corrected any variables pointing to invalid directories. I have put the lines suggested into /etc/profile. Anyone have any ideas? Thanks in advance. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.8/71 - Release Date: 8/12/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk Installations + SER
Ronald Voermans wrote: I'm not sure I understand what you mean... I want to have internal extensions (100, 101, 102, etc.) and some full phone-numbers (10 digits). How do I implement this in *? Ronald Right. We have 3 contexts. 1 is for all incomming traffic from PRI or other carrier. 1 for each company and 1 for all outbound. For us, each company has their own context. This context handles all extensions local to the company. (eg, 100, 101, etc). Then you have a pattern match for when the company dials a 10+ digit number. (or 9 followed by any number of digits) We send all these outbound numbers from all company contexts to a master context for outboud dialing (using Goto). This 'master outbound' context actually includes the 'master incomming' context as part of its dialplan so if any customer dials the 10 digit number of another company, the call stays within asterisk, completly SIP. If no match is found, the call is sent thru a PHP script for least cost routing out via local PRI or SIP carrier. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *confused* - help needed
you can find some resources at the end of this page: http://www.voip-info.org/tiki-index.php?page=ISDN hope it helps. best regards On 8/14/05, Vedran Dakic [EMAIL PROTECTED] wrote: Hello there. What I'm trying to make is - have an asterisk server with sccp/mgcp/skip/h.323 support to handle calls between various company locations. Let's say the company has five different locations, internet connections in each one of them and would like to use it via asterisk to do telephone calls. Hardware-wise I have a couple of dual Opterons and some classic i386 servers (P4). As far as VoIP phones are concerned - I have a couple of Grandstream's, a couple of SwissVoice IP10s's and some Cisco's and Kirk's. I'd like to setup a network that uses those phones (it's not a must to use ALL of them), and to have VoIP telephony between the phones in the company, and if it's not within the company - to use ISDN lines to connect to outsiders. I read something about a possibility to use ISDN modems for these things, but so far I had zero luck configuring this (I have ASUScom's PCI ISDN modem)... Any advices/links? Cheers, Vedran. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RocketVoip?
Does anybody have any experience with setting up Asterisk with the provider RocketVoip? Thanks, // Jonas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conference moderator password
I've been playing around with asterisk for the past few days. One thing which came to mind which could be a useful addition to the meetme functionality would be the possibility to specify a moderator password in meetme.conf. (A moderator in the sense that music is heard until the moderator opens the call and then the call is disconnected when they hangup - options wx). At the moment it seems that the only way to mark a user as a moderator is to use the A option when calling meetme, which means setting up moderator authentication before calling meetme. I think it would be more consistent to include this functionality in meetme and have it read an additional password from meetme.conf Any comments or ideas? Thanks, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Security and SIP
Block sip on a firewall between * and the public internet, and then create rules for your peers IP range. This assumes you know the IP that all peers and client use; if not just block from regions of the world you do not need to connect to/from. We find that most hack attempts come from one well known region, so we block the entire IP range routed to that region. Also, add noload= for the voip protocols you do not use in modules.conf. You are far better off even if you do things like limiting the connections to the ENTIRE ip range of your local Cable/DSL providers. Prevents folks in the rest of the world from even trying to connect. Toll fraud is huge, it looks like you have done the basics, but you should take additional steps many other would call unnecessary since you will get the bill if someone gets it. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of John Fawcett Sent: Monday, August 15, 2005 3:22 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Security and SIP I've now setup SIP for: - internal softphones - registering with external providers (like FWD) for making calls - receiving calls from theese providers For the latter step, it was necessary to forward ports from my NAT to the asterisk server: 5060 + range of ports mentioned in rtp.conf. I was just wondering about how to make this setup as secure as possible. Here's what I've done so far: 1. defined a default context in sip.conf which cannot access any real extension. sip.conf: [general] context=from-unknown-sip extensions.conf: [from-unknown-sip] exten = _.,1,CONGESTION 2. for peers, defined a context which does not provide access to outside lines. sip.conf: [fwd.pulver.com] type=peer username=688426 fromuser=688426 secret=xx host=fwd.pulver.com port=5060 nat=yes canreinvite=no insecure=very context=sip-external disallow=all allow=ulaw 3. for peers, defined insecure=very which should check that the incoming call comes from the same IP as was registered. 4. for internal softphones, which can make outgoing calls, limited registrations to a specific network address using deny/permit sip.conf: [31] type=friend callerid=[EMAIL PROTECTED] 31 host=dynamic deny=0.0.0.0/0.0.0.0 permit=192.168.2.32/255.255.255.255 context=sip-internal secret= disallow=all allow=ulaw allow=alaw Anything else I can do to improve security? I specifically don't want anyone external to be able to make calls. As I've opened port 5060 + rtp.conf ports only for the purpose of receiving calls from services I have registered with, I don't want any external phones to be able to register via this route. Is there any risk of this if someone can guess a password (maybe unlikely but given time this could happen). Thanks, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
On Monday 15 Aug 2005 15:19, Tom Tobias wrote: I am using the correct version of pwlib(1.5.2) and openh323(1.12.2) for the stable asterisk build. Both packages configure and compile with no problems. However when compiling chan_h323 from the asterisksource/channels/h323 directory I get this error. It's not an error, it's a warning and it can be ignored. Chan-h323.h:31: warning; ‘sockaddr_in bindaddr’ defined but not used arc r libchanh323.a ast_h323.o There have been posts of similar messages but none with the specific syntax as the one above. Those posts have mentioned commenting some part of the source code in order to build. I have combed the Makefile and corrected any variables pointing to invalid directories. I have put the lines suggested into /etc/profile. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_sccp and dynamic DNS
Hi, everybody. I was just running into a DynDNS Problem. One of our phones has a fixed IP, but the * box only has a dynDNS IP and a fixed name. So all phones running SIP are able to survive the change of the IP address, but not our Cisco 7920 - it changes state to connecting to CM0 and hangs there until reboot. After reboot it looks up the current IP address of the * box and works fine for the next 24 hours. My question now is: Is there an option in sccp.conf or somewhere else to tell the 7920 to do DNS lookups once in a while so it updates the IP address dynamically? Any help there would be great! Thanks in advance. Armin Lediger -- HotZone GmbH Würzburg - schnurlos glücklich! WLAN und VoIP Dienstleistungen Arndtstr. 5 97072 Würzburg [EMAIL PROTECTED] Voice 0931-2056064 Fax 0931-2056063 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_sccp and dynamic DNS
Armin Lediger wrote: Hi, everybody. I was just running into a DynDNS Problem. One of our phones has a fixed IP, but the * box only has a dynDNS IP and a fixed name. So all phones running SIP are able to survive the change of the IP address, but not our Cisco 7920 - it changes state to connecting to CM0 and hangs there until reboot. After reboot it looks up the current IP address of the * box and works fine for the next 24 hours. My question now is: Is there an option in sccp.conf or somewhere else to tell the 7920 to do DNS lookups once in a while so it updates the IP address dynamically? Why don't you chuck a network card in the box and connect to the internal side? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] User in two queues receive two calls at once
Hi all, I have several queues defined in Asterisk, but when I put a user in two queues, the user (can) get two calls at once. (One from each queue). Right now, I configured some phones to only accept one call at once, but that has the side-effect, that also direct calls won't come through. Is there some configuration option to make sure that if a user has an open call from a queue, that he/she will not get a call from that- or another queue (until the call from the first queue is finished) ? I have thought of dynamically removing the user from all queues when he/she receives an incoming queue-call, but wouldn't prefer that route.. thanks, Leon de Rooij [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Cisco and protocol application invalid
Hello, thanks for the replies so far. In regards to the below quoted answer: If the phone locks up before SIP firmware has booted, then there's absolutely no way to set these settings manually from the phone itself. The situation is simply that I have no idea which tftp settings that are already in the phone, else I would have just changed the lan and set up a tftp server on the IP address the phone is searching for. Tried console, but got no connection. The cable was home made, I could have gone wrong somewhere, or the scheme used can be wrong - dunno. So as for now I am kinda lost. Bjorn -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av nkm Sendt: 15. august 2005 11:22 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Cisco and protocol application invalid On 8/14/05, Bjorn Ove Kristiansen [EMAIL PROTECTED] wrote: Is there any way of getting to know which IP address Cisco uses to contact TFTP? Why you're making things hard for yourself for no good reason? Unlock the config, put a static entry for ip that belongs to the segment its sitting on (RFC1918 or not), put a static entry for your TFTP server and upgrade. When all goes well setup your DHCP pools with the tftp options for future use. Regards, Bjorn Hope that helps a bit nkm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.9/72 - Release Date: 14.08.2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.9/72 - Release Date: 14.08.2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Cisco and protocol application invalid
Hello! The issue is simply that I don't know which IP address the phone tries to connect to. I am not very familiar with dhcpd (never put it up by hand), so I'm not sure how the below would help me, but from what I can tell, I still need information on which IP-address the phone is trying to find its tftp on, right? Bjorn -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Tony Hoyle Sendt: 14. august 2005 23:23 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Cisco and protocol application invalid Michiel van Baak wrote: I have put this in my dhcpd.conf to make sure my cisco phones connect to my TFTP server: server-name 192.168.2.1; I'd be surprised if that worked... the server name is for.. um.. the name of the server :) Try: option tftp-boot-server code 150 = ip-address; option tftp-boot-server 192.168.44.3; Or you can hardcode it in the alternative TFTP server setting on the phone itself. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.8/71 - Release Date: 12.08.2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.9/72 - Release Date: 14.08.2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] MS Live Communications Server
Search google with sip pstn site:www.microsoft.com
You will find out how to configure LCS static routing to SIP Gateway,
like Asterisk
but you need patch Asterisk to support TCP.
http://bugs.digium.com/view.php?id=4903
Step1: configure LCS 2005 to let sip uri: [EMAIL PROTECTED] to route to
next hop: pstngw ip address
Step2: patch your asterisk chan_sip.c to support TCP
Step3: configure your Asterisk sip.conf, extensions.conf
simple example :-)
sip.conf
context=sip_incoming
extensions.conf
[sip_incoming]
exten = _XX.,1,Answer
exten = _XX.,2,Noop(do trust ip check or some authentication)
exten = _XX.,3,Dial(Zap/${EXTEN}SIP/${EXTEN})
I still find out how to let LCS 2005 accept SIP invite from Asterisk,
Need more help.
2005/8/13, bubuk [EMAIL PROTECTED]:
Hi,
I already posted this in the user list, but this list is probably the
better one.
My question was: Does anyone played around with the LCS and Asterisk?
Because the LCS is doing no RFC compliant SIP, i wonder if it can work.
Google couldn't tell me. If someon heard about that, please let me know.
Thank you
Volker
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Re: [Asterisk-Users] User in two queues receive two calls at once
Register with two different accounts from the same phone. One for Queue Calls, and one for direct calls. Greetz, Zoa. --- Asterisk tutorials: http://www.asteriskguru.com --- Leon de Rooij wrote: Hi all, I have several queues defined in Asterisk, but when I put a user in two queues, the user (can) get two calls at once. (One from each queue). Right now, I configured some phones to only accept one call at once, but that has the side-effect, that also direct calls won't come through. Is there some configuration option to make sure that if a user has an open call from a queue, that he/she will not get a call from that- or another queue (until the call from the first queue is finished) ? I have thought of dynamically removing the user from all queues when he/she receives an incoming queue-call, but wouldn't prefer that route.. thanks, Leon de Rooij [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Cisco and protocol application invalid
On Mon, 2005-08-15 at 17:28 +0200, Bjørn Ove Kristiansen wrote: Hello! The issue is simply that I don't know which IP address the phone tries to connect to. I am not very familiar with dhcpd (never put it up by hand), so I'm not sure how the below would help me, but from what I can tell, I still need information on which IP-address the phone is trying to find its tftp on, right? Hook the phone directly with a crossover cable to a linux box. Run tcpdump, and see what it needs/wants. And then give it that :) You might need to have dhcp and tftpserver installed locally and running to get it happy. But you could set that port to the ip it is looking for, etc etc. I have seen this error message when the phone is give P00xx instead of the P0S or vice versa. -- respectfully, Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
A red alarm means I don't see any signal. A blue alarm means I see a signal and something downstream (repeater etc) is saying they don't see a signal. I used this as my reference: http://www.fratec.com/FAQ/NFO/NFO_WAN_009.HTML Slips sometimes cause an LOF condition, sometimes they don't. At least when I worked NorTel DMS250 switches back in the 80's that's the way it was. Thanks for making me stretch my mind! So I had an old Digium wildcard sitting around that I put inplace of the new TE110P and the Blue Alarms seem to have cleared. There are still SLIP errors and the call quality still sounds bad compared to the SIP only calls. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 registration problem
iam new to h323 ,i just download it and compile and it gives me these messages on console : Sep 13 00:20:30 WARNING[15443]: chan_oh323.c:4014 oh323_gk_check: Gatekeeper discovery failed. -- Retrying gatekeeper registration. does any body knows whats going on. Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] v92 modems
I got one from www.broad-tel.com. It works fine. On 8/12/05, Douglas Logan [EMAIL PROTECTED] wrote: Yes, but your results may vary. Apparently some people have problems with clone cards (aka regular modems), dropping calls, and having echos. (Then again some people have reported no problems at all). E-bay is a good source for these. You can also check out this list with more information about Asterisk clone cards here: http://www.voip-info.org/tiki-index.php?page=X100P+clone On 8/12/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Friday 12 August 2005 06:42, [EMAIL PROTECTED] wrote: Hello, Is it possible to use v92 ( a few chipsets version ) modem as FXO PCI modules ? Short answer: no. Longer answer: perhaps, but you're on your own. Your googling efforts should have shown you that. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] User in two queues receive two calls at once
Hi, Hmm.. didn't think of that.. Thanks for the tip.. Though it wouldn't really work as it's not possible to set CWI (Call Waiting Indication) seperately for each logged-in user.. (We currently use SNOM190 and Cisco7960 phones).. Also it would change our entire system (database, webinterface and all).. Isn't there another option ? thanks again, Leon de Rooij [EMAIL PROTECTED] On Mon, 2005-08-15 at 18:39 +0300, Zoa wrote: Register with two different accounts from the same phone. One for Queue Calls, and one for direct calls. Greetz, Zoa. --- Asterisk tutorials: http://www.asteriskguru.com --- Leon de Rooij wrote: Hi all, I have several queues defined in Asterisk, but when I put a user in two queues, the user (can) get two calls at once. (One from each queue). Right now, I configured some phones to only accept one call at once, but that has the side-effect, that also direct calls won't come through. Is there some configuration option to make sure that if a user has an open call from a queue, that he/she will not get a call from that- or another queue (until the call from the first queue is finished) ? I have thought of dynamically removing the user from all queues when he/she receives an incoming queue-call, but wouldn't prefer that route.. thanks, Leon de Rooij [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No translator path exists for channel type MGCP Comfort noise support incomplete
ONLY ON MONDAY!
Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this?
*CLI -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: DownAug 15 11:20:19 NOTICE[13883]: rtp.c:284 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.0.241 -- Executing Dial("MGCP/aaln/[EMAIL PROTECTED]", "MGCP/aaln/[EMAIL PROTECTED]") in new stackAug 15 11:20:19 WARNING[13883]: channel.c:2175 ast_request: No translator path exists for channel type MGCP (native 4) to 256Aug 15 11:20:19 NOTICE[13883]: app_dial.c:1091 dial_exec_full: Unable to create channel of type 'MGCP' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'MGCP/aaln/[EMAIL PROTECTED]'
status
is 'CHANUNAVAIL'
*CLI
here is my mgcp.conf and extensions.conf
IPD:/etc/asterisk# more extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
[extensions]
exten = 123,1,Dial,MGCP/aaln/[EMAIL PROTECTED]
exten = 124,1,Dial,MGCP/aaln/[EMAIL PROTECTED]
[directdial]
ignorepat = 9
exten = 9,1,MGCP/aaln/[EMAIL PROTECTED]
exten = 9,2,Congestion
[international]
ignorepat = 9
exten = _9011.,1,Dial(Zap/g2/${EXTEN:1})
exten = _9011.,2,Congestion
include = longdistance
[longdistance]
ignorepat = 9
exten = _91NXXNXX,1,Dial(Zap/g2/${EXTEN:1})
exten = _91NXXNXX,2,Congestion
include = local
[local]
ignorepat = 9
exten = _9NXXNXX,1,Dial,MGCP/aaln/[EMAIL PROTECTED]
exten = _9NXXNXX,2,Congestion
[outbound-default]
include = extensions
include = directdial
include = longdistance
include = local
and the *** mgcp.conf
IPD:/etc/asterisk# more mgcp.conf
; MGCP Configuration for Asterisk
[general]
port=2727
bindaddr=0.0.0.0
allow=ulaw
allow=g729
allow=g726
tos=0x85
srvlookup=yes
wcardep=aaln/*
; Bob's CMG #1
[192.168.0.241]
context=outbound-default
host=192.168.0.241
wcardep=*
line = *
;
; Line 1
;
callerid = "John" 123
callgroup=0
pickupgroup=0
nat=no
threewaycalling=yes
transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer
callwaiting=yes ; this might be a cause of trouble for ip10s
cancallforward=yes
line = aaln/1
;
; Line 2
;
callerid = "Jane" 124
callgroup=0
pickupgroup=0
nat=no
threewaycalling=yes
transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer
callwaiting=yes ; this might be a cause of trouble for ip10s
cancallforward=yes
line = aaln/2
;outbound t1 port 5-6 cross connected to cmg 6:1:1:3-4
line = aaln/3
line = aaln/4
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Re: [Asterisk-Users] No translator path exists for channel type MGCP Comfort noise support incomplete
On 09:38, Mon 15 Aug 05, kurt turner wrote: ONLY ON MONDAY! Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this? rant That's what you get from trashing Debian in favour of RedHat /rant Please don't take this message seriously ;) Just couldn't resist. Sorry -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs order
Hi, asterisk will negotiate codecs for both parties independently (use sip show peer peer and look for codec order entry), so, if you have prefered codec g729 for your sip phone/peer, asterisk will use them (regardles of codec setting for other party - if codecs does not match, asterisk will try to transcode between) imho ;-) PJ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No translator path exists for channel type MGCP Comfort noise support incomplete
no translator path = no codec... Have you called Digium to get your G729 (format 256)codecs released and re-registered them to the new(ly configured) box??? At the cli, type show g729... does it give an error or show g729 information? regards, Derek - Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, August 15, 2005 10:50 AM Subject: Re: [Asterisk-Users] No translator path exists for channel type MGCP Comfort noise support incomplete On 09:38, Mon 15 Aug 05, kurt turner wrote: ONLY ON MONDAY! Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this? rant That's what you get from trashing Debian in favour of RedHat /rant Please don't take this message seriously ;) Just couldn't resist. Sorry -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs order
Pavel Jezek wrote: Hi, asterisk will negotiate codecs for both parties independently (use sip show peer peer and look for codec order entry), so, if you have prefered codec g729 for your sip phone/peer, asterisk will use them (regardles of codec setting for other party - if codecs does not match, asterisk will try to transcode between) imho ;-) It does seem to be a weakness of asterisk.. it's creating load on the server when it doesn't need to. Really it should look at the capabilities of both ends and see if there's a common set, and only start transcoding if there's no overlap. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Switch between FXS ports
Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch between FXS ports
Innocent Evil wrote: Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. First, FXS = handset / FXO = telco line. Second, you don't. Does the telephone company let you do this now, if so, how - otherwise, no you can't. Chris -- Christopher L. Wade ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BRI Hunting, using both channels on one msn
Hello All, Has anyone configured bri to answer for only one msn? In essence, when the primary is busy I want to have channel 2 ring. I am using an eicon diva server bri I know I saw it in the windows interface, but don't see it in the linux setup. Regards, Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Switch between FXS ports
As far as I remember, you can't really do that (because the telco isn't switching the call), what you'll want to do is have a hunt group set up --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Innocent Evil -Sent: Monday, August 15, 2005 2:17 PM -To: asterisk-users@lists.digium.com -Subject: [Asterisk-Users] Switch between FXS ports - -Hello, - -I have two FXS port on my TDM card. -channel 4 is attached with a telco line that I use -frequently. And channel 3 have another telco line. but I dont -publish that number to my friends. -If I receive a call through channel 4, how can I handover -that call to channel 3 ..so that I can keep channel 4 open -for incoming call. - -Thanks,___ -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch between FXS ports
Chris Wade wrote: Innocent Evil wrote: Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. First, FXS = handset / FXO = telco line. Second, you don't. Does the telephone company let you do this now, if so, how - otherwise, no you can't. You'd have to get the telco to set up a redirect. But then you wouldn't receive a call on chan 4... You can probably do it with ISDN, but probably not with PSTN... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch between FXS ports
Innocent Evil wrote: Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Basically, you'd need to have the telco have the phone calls auto forwarded to the next available line. That's pretty common for them to do. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch between FXS ports
Sorry for the typo. Do I need to ask my telco, if I want to use Asterisk as a PBX in a home/small biz/large biz and I want one hunting number. Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Mon, 15 Aug 2005 13:20:17 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Switch between FXS ports Innocent Evil wrote: Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. First, FXS = handset / FXO = telco line. Second, you don't. Does the telephone company let you do this now, if so, how - otherwise, no you can't. Chris -- Christopher L. Wade ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch between FXS ports
On Mon, 2005-08-15 at 13:20 -0500, Chris Wade wrote: First, FXS = handset / FXO = telco line. Ditto this. Maybe something like fax-callback; call-in, hangup, Asterisk dials back on the other channel using the CID received - a purely physical solution. Otherwise, have the telco setup a rotary hunt to go between the two lines. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Switch between FXS ports
As far as I remember, you can't really do that (because the telco isn't switching the call), what you'll want to do is have a hunt group set up Yesss... this is exactly I am looking for. How can I do that? Thanks, --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Innocent Evil -Sent: Monday, August 15, 2005 2:17 PM -To: asterisk-users@lists.digium.com -Subject: [Asterisk-Users] Switch between FXS ports - -Hello, - -I have two FXS port on my TDM card. -channel 4 is attached with a telco line that I use -frequently. And channel 3 have another telco line. but I dont -publish that number to my friends. -If I receive a call through channel 4, how can I handover -that call to channel 3 ..so that I can keep channel 4 open -for incoming call. - http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Switch between FXS ports
Yes, you need to ask the telco to autoforward your chan 4 num to chan 3 (called hunt grouping), there may be a fee. Also not sure if that's available for a standard residential line (or just POTS in general). You don't need to tell them why, just tell 'em you want it. No need to confuse 'em. Sherwood McGowan --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Innocent Evil -Sent: Monday, August 15, 2005 2:29 PM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: [Asterisk-Users] Switch between FXS ports - -Sorry for the typo. - -Do I need to ask my telco, if I want to use Asterisk as a PBX -in a home/small biz/large biz and I want one hunting number. - -Thanks, - - - -Original Message- - From: [EMAIL PROTECTED] - Sent: Mon, 15 Aug 2005 13:20:17 -0500 - To: asterisk-users@lists.digium.com - Subject: Re: [Asterisk-Users] Switch between FXS ports - - Innocent Evil wrote: - Hello, - - I have two FXS port on my TDM card. - channel 4 is attached with a telco line that I use frequently. And - channel 3 - have another telco line. but I dont publish that number -to my friends. - If I receive a call through channel 4, how can I handover -that call - to channel 3 ..so that I can keep channel 4 open for -incoming call. - - First, FXS = handset / FXO = telco line. Second, you -don't. Does the - telephone company let you do this now, if so, how - -otherwise, no you - can't. - - Chris - - -- - Christopher L. Wade - - ___ - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: - - -http://lists.digium.com/mailman/listinfo/asterisk-users___ - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch between FXS ports
I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Basically, you'd need to have the telco have the phone calls auto forwarded to the next available line. That's pretty common for them to do. That's exactly what I do with our business line. Call Forward on Busy is a common description for that telco service. (I simply forward that next call to an unlisted/unpublished number which also terminates in Asterisk.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with sound device
I am getting this whenever I start asterisk. Aug 15 15:03:58 WARNING[11754] chan_oss.c: Read error on sound device: Resource temporarily unavailable This is my sound card: Multimedia audio controller: Fortemedia, Inc Xwave QS3000A I am not sure... what I am doing wrong. Please help. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dell Poweredge 1400
I think this email got mixed with other emails thks. Hi all, In this moment I have the opportunity to install asterisk in Poweredge 1400 Dell Server (PIII, 2 GB of RAM). I wonder if any of you have any experience running asterisk (+ Digium cards) on this kind of hardware, any comment about know problems or good experiences are welcome. Thanks in advance. Alejandro Acosta,- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Switch between FXS ports
I am clear with this issue. Thanks everybody for answering me. -Original Message- From: [EMAIL PROTECTED] Sent: Mon, 15 Aug 2005 10:16:34 -0800 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Switch between FXS ports Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch between FXS ports
The Call Forward On Busy does cost YOU money each time you forward a call. Call hunting group is different from Call forwarding. In a hunt group you have 2 or more phone lines grouped together. When a call for a number associated with the group comes into the telco switch, the switch checks which lines inside the group are available, then the switch selects one of the available lines where it will send the call to. This selection is done using a predefined algorythm (random, round robin, ascending, descending,) Call hunting groups are also in most times used on a T1 PRI or E1 PRI. When a call comes in for a phonenumber associated with the T1/E1 only 1 channel will ring. Some telcos may charge additional fees to setup a call hunting group, but in cases you make a certain usage of Call forwarding, it may be less expensive to use a call hunting group. Best regards, Marc Rich Adamson wrote: I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Basically, you'd need to have the telco have the phone calls auto forwarded to the next available line. That's pretty common for them to do. That's exactly what I do with our business line. Call Forward on Busy is a common description for that telco service. (I simply forward that next call to an unlisted/unpublished number which also terminates in Asterisk.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell Poweredge 1400
Alejandro... Go search the archive... There are tons of posts regarding Dell equipment Here is how to do so if you do not know... Go to www.google.com Enter the following... site:lists.digium.com Dell Poweredge Thanks, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Acosta Sent: Monday, August 15, 2005 12:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Dell Poweredge 1400 I think this email got mixed with other emails thks. Hi all, In this moment I have the opportunity to install asterisk in Poweredge 1400 Dell Server (PIII, 2 GB of RAM). I wonder if any of you have any experience running asterisk (+ Digium cards) on this kind of hardware, any comment about know problems or good experiences are welcome. Thanks in advance. Alejandro Acosta,- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P problem
On Fri, Aug 12, 2005 at 09:03:42PM -0500, Tim Connolly wrote: I'm having lots of stability problems with my 411's. I'm not blaming the 411 yet, just seems odd that I ran for months on a TE110P with a peak of 10-15 calls, and now my box kernel panics each time it hits the same load. Granted, its got 4 PRI's now, but still only 10-15 calls will kill it. Does seem to kill the echo as long as the zttest comes back clean. Right now, we're trying to work out some issues that we have seen in customers machines similar to this. If the kernel panics are caused by the TE411P driver (wct4xxp) then you might want to try calling Digium tech support about this so that we can help you get it fixed. -- Matthew Fredrickson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo calibration with ztmonitor and a test line from a telco
Hello everyone, Does anyone have experience with echo calibration for TDM card with rxgain and txgain (Zapata.conf) and ztmonitor (CVS Head)? I have found very few information about it and what I have found makes me confused. I have a phone number provided by my TelCo(1004 hz at 0db) and from what I saw, I am supposed to calibrate my rxgain to get a 14800 value with ztmonitor . Here is the information I found: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.ht ml Does anyone have successfully reduced echo with this procedure? My main problem is that when I get 14800 with ztmonitor, I have now a rxgain=14 and it seem to be too high for asterisk and I cannot dial out anymore. Any suggestions? Thanks in advance for your pointers Regards Ken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch between FXS ports
The Call Forward On Busy does cost YOU money each time you forward a call. No, that is telco dependent. Most US telco's do not charge for that as long as the forwarded number is a local number. If its not, then LD charges apply. But in some US cities, you are correct that an additional change _might_ apply. Call hunting group is different from Call forwarding. Yup, that will work as well, however in the US that generally involves a Service Order (and charge), and in some cases will require a telephone number change (depends upon the exact type of central office switch the telco is using). Rich In a hunt group you have 2 or more phone lines grouped together. When a call for a number associated with the group comes into the telco switch, the switch checks which lines inside the group are available, then the switch selects one of the available lines where it will send the call to. This selection is done using a predefined algorythm (random, round robin, ascending, descending,) Call hunting groups are also in most times used on a T1 PRI or E1 PRI. When a call comes in for a phonenumber associated with the T1/E1 only 1 channel will ring. Some telcos may charge additional fees to setup a call hunting group, but in cases you make a certain usage of Call forwarding, it may be less expensive to use a call hunting group. Best regards, Marc Rich Adamson wrote: I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Basically, you'd need to have the telco have the phone calls auto forwarded to the next available line. That's pretty common for them to do. That's exactly what I do with our business line. Call Forward on Busy is a common description for that telco service. (I simply forward that next call to an unlisted/unpublished number which also terminates in Asterisk.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax Issues
I have a user who has a fax machine plugged into an ATA. They are able to SEND faxes just fine. Faxes go through wonderfully. However, when someone tries to send them a fax, their fax machine never receives it. And eventually the sending machine just errors out. Any thoughts? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo calibration with ztmonitor and a test line from a telco
On 8/15/05, Ken Dresdell [EMAIL PROTECTED] wrote: Hello everyone, Does anyone have experience with echo calibration for TDM card with rxgain and txgain (Zapata.conf) and ztmonitor (CVS Head)? I have found very few information about it and what I have found makes me confused. I have a phone number provided by my TelCo(1004 hz at 0db) and from what I saw, I am supposed to calibrate my rxgain to get a 14800 value with ztmonitor . Here is the information I found: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.ht ml Does anyone have successfully reduced echo with this procedure? My main problem is that when I get 14800 with ztmonitor, I have now a rxgain=14 and it seem to be too high for asterisk and I cannot dial out anymore. Any suggestions? Thanks in advance for your pointers Regards Ken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have been doing a bit of this too lately. This was also useful. http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRI Hunting, using both channels on one msn
On Mon, 15 Aug 2005 [EMAIL PROTECTED] wrote: Hello All, Has anyone configured bri to answer for only one msn? In essence, when the primary is busy I want to have channel 2 ring. I am using an eicon diva server bri I know I saw it in the windows interface, but don't see it in the linux setup. This is normal behaviour. What exactly is your problem? Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Now that I've looked back over my work for the past few days I realize that I was trying to play with the txgain/rxgain to adjust the levels and hope to smooth out the line noise. Well, any integer other than zero for either of those values causes BLUE alarms and all the channels to reset in Asterisk. SO, my problem now is related to SLIP errors. I still have all the same line noise as I have but the only errors I am seeing are Slip error. PS: Thanks for all the help so far. Thanks, Geoff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch between FXS ports
On Mon, August 15, 2005 3:50 pm, Rich Adamson said: That's exactly what I do with our business line. Call Forward on Busy is a common description for that telco service. (I simply forward that next call to an unlisted/unpublished number which also terminates in Asterisk.) In my very limited experience, Call Forward on Busy will only work once. In my situation, I have two POTS lines from the telco and a VoIP service provider. I had the telco enable Call forward on Busy so Line-1 rolled over to Line-2 and Line-2 rolled over to the VoIP carrier. I setup a group of users to call in on the number for Line-1. - The first caller go in on Line-1 - The second caller got in on Line-2 - the others got busy signals. This may just be my telco. Paul -- Paul Dugas, Computer Engineer Dugas Enterprises, LLC [EMAIL PROTECTED] phone: 404-932-1355 522 Black Canyon Park http://dugas.cc fax: 866-751-6494 Canton, GA 30114 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with FWD connection rejected
Sean Rima wrote: Using the install instructions for [EMAIL PROTECTED], I setup a FWD account, this I tested using X-Lite and it works okay, Nowever I cannot make calls to fwd using Asterisk, my log showes: Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected: Registration Refused Well folks, I did it, sorted it out all on my ownsome :P) I did not enable the option at FWD to allow me to use IAX, once I enabled it, it worked :) Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie FreeWorldDial: 689482 smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to remove standard ISDN drivers from RedHat
I have newly installed a RedHat 4.0 EL rebuild. The install was done without the ISDN card present. After disabling kudzu and haldaemon I inserted the card. Stil that *($^%$($^!! kudzu shit modified my config and is loading hisax, crc_ccit and isn modules. Even worse, they do not appear in /etc/modprobe.conf which means that that f*cking kudzu added the modules to initrd. I have googled for hours and browsed through all the redhat docs but I cannot find how to remove these modules. All the docs mention is to 'simply comment them out from modprobe.conf' well, they aren't there. Does anybody know how I can remove these modules? They are really a pain in the ass because now I cannot start asterisk from the init scripts only from rc.local Thanx a 1,000,000 :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Firewall will definatelyincrease jittersinyourvoice conversation
Typically a hardware firewall is specialized and uses ASICs. Because the solution utilizes specialized chips tailored to the task, this is considered a hardware based solution. Of course software is involved but it too is specialized and is even proprietary in nature. A software firewall, be it BlackICE or even a Linux on PC uses no specialized hardware. Thus the software designation. It runs on pretty much any x86 hardware (Linux at least) and is not proprietary in nature. That is the general meaning when people say hardware or software firewall. Sure, both technically use some form of hardware and software. But the specialization of that hardware is what makes it designated as hardware based or software based. There have been countless arguments over firewalls in the software vs. hardware arena. At this point and time, I can say I feel that both have great purpose and functionality. I prefer my Pix because I use VPN tunnels to certain sites that have Cisco on the other side and it makes things easier. The configuration of my firewall is also very simplified with my Pix. I ran a Linux firewall for quite a while and I loved it. With the amount of power available to the modern (or even somewhat outdated) PC, you can leverage plenty of performance out of a marginal box. So, to each there own! Use what works best for you application. Great points on single entry point being easier BTW. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Travers Sent: Saturday, August 13, 2005 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Firewall will definatelyincrease jittersinyourvoice conversation Wiley Siler wrote: The question was not can I secure a Linux box without a hardware firewall. The question (or statement really) was will a firewall add jitter and lower performance. A good firewall architecture w/QoS will actually prevent jitter and increase performance, I might add. That answer is obviously a big NO. Can you secure a Linux (or even Windows) machine by closing ports? Sure. It helps immensely. However, an advantage of hardware is that you are physically separating the traffic from the end point. The analogy I would use here is that you could purchase a safe for each person in your house and have them each keep all their valuables in it, but it is often cheaper and easier to focus on securing entrence-points. The same is doubly true for office buildings, and also quite true for computer networks. I typically use used P1's running Linux for firewalls. They work great and have all the capabilities I need including QoS and secure management. Sure, all the ports closed on a Linux box can protect that machine. However, having only web (for example) traffic going to your Apache server is really beneficial. The server can focus on delivering pages and not spend any CPU cycles on is this a good packet? Should I drop it?. A firewall (software or hardware) should also be able to better deal with DOS and things of that nature. Port securing does nothing to assist with DOS. DOS doesn't include a TCP/IP stack does it? ;-) By Things of that nature are you including CP/M? Actually port securing can provide some measure of protection against DoS attacks in that fewer services are available to attack. However, you are correct that this protection is probably insignificant. So... You are totally right, you can secure a box that way. However, a firewall (be it software or hardware) is far superior a method. When you say software or hardware I assume you mean hardware like PIX and software like BlackIce. I am not sure where a stripped down Linux version running on a P1 which does firewalling and only firewalling fits in. I call that type of system a hardware firewall simply because it is a dedicated piece of hardware which does perimiter control and only perimiter control. Where VOIP is concerned, use a dedicated firewall system with QoS capabilities. Period. (Yes it is possible to run such a system on Windows, but I certainly don't advise it.) I prefer the hardware method myself as it is a matter of management and additional features. However, for some, a software method may be better. I ran Mandrake SNF (a shorewall implementation) for a long time so I have been there. Considering you can run a Linux firewall on a 386 machine worth $20 makes the fact that so many people don't have firewalls seem just ridiculous. Bear in mind that finding replacement parts (NIC's etc) for your 386 may not be trivial. That is why I use P1's with PCI slots... Also it is often impossible to get OpenGK to compile on such a machine due to memory limitations (my P1 firewall even has this problem and it has a whopping 32MB RAM). So the older you go, the less functionality you may be able to add. Best Wishes, Chris Travers Metatron Technology
[Asterisk-Users] Only single channel recorded with Monitor
We are using the following to record conversations.
exten = _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP})
exten = _1XXX.,2,Monitor(wav,${CALLFILENAME},m)
exten = _1XXX.,3,Dial(IAX2/4506:[EMAIL PROTECTED]/${EXTEN:1})
exten = _1XXX.,4,Congestion
exten = _1XXX.,104,Congestion
This was working previously to record both sides of the
conversation but now we only have the initiating caller channel
being recorded. Occasionaly the other caller is also recorded
but the speed of the recording is completely wrong causing
distortion and out of sync.
Here fwiw are the logs.
Aug 15 18:31:32 DEBUG[9995]: build_route: Contact hop: sip:[EMAIL
PROTECTED]:5060;line=ikojqrcx
Aug 15 18:31:32 DEBUG[9995]: Device 'SIP/snom' changed to state '2'
Aug 15 18:31:32 VERBOSE[9995]: -- Executing SetVar(SIP/snom-7214,
CALLFILENAME=call_to_00NUMBER_HIDDEN_dated_20050815-183132) in new stack
Aug 15 18:31:32 VERBOSE[9995]: -- Executing Monitor(SIP/snom-7214,
wav|call_to_00NUMBER_HIDDEN_dated_20050815-183132|m) in new stack
Aug 15 18:31:32 VERBOSE[9995]: -- Executing Dial(SIP/snom-7214,
IAX2/4506:[EMAIL PROTECTED]/00NUMBER_HIDDEN) in new stack
Aug 15 18:31:32 VERBOSE[9995]: -- Called 4506:[EMAIL
PROTECTED]/00NUMBER_HIDDEN
Aug 15 18:31:32 DEBUG[9995]: Device 'IAX2/4506/2' changed to state '2'
Aug 15 18:31:32 VERBOSE[9995]: -- Call accepted by 80.127.191.55 (format
G729A)
Aug 15 18:31:32 VERBOSE[9995]: -- Format for call is G729A
Aug 15 18:31:34 VERBOSE[9995]: -- IAX2/4506/2 is ringing
Aug 15 18:31:34 DEBUG[9995]: Ooh, voice format changed to 256
Aug 15 18:31:34 DEBUG[9995]: Ooh, format changed from UNKN to G729A
Aug 15 18:31:45 VERBOSE[9995]: -- IAX2/4506/2 stopped sounds
Aug 15 18:31:45 VERBOSE[9995]: -- IAX2/4506/2 answered SIP/snom-7214
Aug 15 18:31:45 DEBUG[9995]: Stopping retransmission on '[EMAIL PROTECTED]' of
Response 2: Found
Any ideas how to fix this?
Thanks
--
Eric Smith
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RE: [Asterisk-Users] Firewall will definatelyincreasejitters inyourvoice conversation
Do you mean this occurs when traffic is passed over an IPSec tunnel or that it occurs anytime a tunnel is use on a machine that also is passing VoIP traffic (outside the tunnel)? I assume you must mean over the tunnel but I am curious... Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Saturday, August 13, 2005 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Firewall will definatelyincreasejitters inyourvoice conversation On that note... IPSec tunnels seem to reek havoc on the echo canceling/training process. Anytime our Cisco PIX loads up, the echo complaints start coming in. Stay away from the IPSec tunnels. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Travers Sent: Saturday, August 13, 2005 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Firewall will definately increasejitters inyourvoice conversation Rich Adamson wrote: That's a crack of crap sold by the marketing (not sales) people selling firewalls. If you know what you're doing, one can very easily secure any linux system to function on the Internet (etc) without a firewall. It all depends on your level of knowledge/skills on how to disable those items that are not really needed in your environment. Start with a 'netstat -a' to identify those ports that are listening, and shut those items down that you don't want exposed. You can do the same for any MS system as well. But you still want a firewall here especially if you have several VOIP systems which could be making independent connections to the internet. The firewall in this case will hopefully not only do things like VPN for securing your data in trasit between your office and a remote one, but it will also provide a platform for QoS/traffic shaping. To avoid the firewall here is actually *asking* for sound quality problems in addition to the fact that you no longer have the entrence point to your network secured. Now to your point Almost any Linux system can be configured (if you know what you are doing) to perform all these firewalling functions. Just add an extra network card, put it on the perimeter of your network, set up iptables, traffic shaping, uninstall unnecessary software, use Netstat to doublecheck listening ports, etc. and you have your firewall. A firewall doesn't have to be expensive but some form of perimiter control is very helpful in these cases. Best Wishes, Chris Travers Metatron Technology Consulting ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Only single channel recorded with Monitor
Try reinstalling sox - it is responsible for mixing the caller and
callee channels. Also, if IAX2/4506:[EMAIL PROTECTED] is your
real username and password, change them asap, you just made it available
to 1+ people and the archives ;)
Regards,
Vahan
Eric Smith wrote:
We are using the following to record conversations.
exten = _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP})
exten = _1XXX.,2,Monitor(wav,${CALLFILENAME},m)
exten = _1XXX.,3,Dial(IAX2/4506:[EMAIL PROTECTED]/${EXTEN:1})
exten = _1XXX.,4,Congestion
exten = _1XXX.,104,Congestion
This was working previously to record both sides of the
conversation but now we only have the initiating caller channel
being recorded. Occasionaly the other caller is also recorded
but the speed of the recording is completely wrong causing
distortion and out of sync.
Here fwiw are the logs.
Aug 15 18:31:32 DEBUG[9995]: build_route: Contact hop: sip:[EMAIL
PROTECTED]:5060;line=ikojqrcx
Aug 15 18:31:32 DEBUG[9995]: Device 'SIP/snom' changed to state '2'
Aug 15 18:31:32 VERBOSE[9995]: -- Executing SetVar(SIP/snom-7214,
CALLFILENAME=call_to_00NUMBER_HIDDEN_dated_20050815-183132) in new stack
Aug 15 18:31:32 VERBOSE[9995]: -- Executing Monitor(SIP/snom-7214,
wav|call_to_00NUMBER_HIDDEN_dated_20050815-183132|m) in new stack
Aug 15 18:31:32 VERBOSE[9995]: -- Executing Dial(SIP/snom-7214,
IAX2/4506:[EMAIL PROTECTED]/00NUMBER_HIDDEN) in new stack
Aug 15 18:31:32 VERBOSE[9995]: -- Called 4506:[EMAIL
PROTECTED]/00NUMBER_HIDDEN
Aug 15 18:31:32 DEBUG[9995]: Device 'IAX2/4506/2' changed to state '2'
Aug 15 18:31:32 VERBOSE[9995]: -- Call accepted by 80.127.191.55 (format
G729A)
Aug 15 18:31:32 VERBOSE[9995]: -- Format for call is G729A
Aug 15 18:31:34 VERBOSE[9995]: -- IAX2/4506/2 is ringing
Aug 15 18:31:34 DEBUG[9995]: Ooh, voice format changed to 256
Aug 15 18:31:34 DEBUG[9995]: Ooh, format changed from UNKN to G729A
Aug 15 18:31:45 VERBOSE[9995]: -- IAX2/4506/2 stopped sounds
Aug 15 18:31:45 VERBOSE[9995]: -- IAX2/4506/2 answered SIP/snom-7214
Aug 15 18:31:45 DEBUG[9995]: Stopping retransmission on '[EMAIL PROTECTED]' of
Response 2: Found
Any ideas how to fix this?
Thanks
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research Development dept.
adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
url:http://www.arminco.com/
version:2.1
end:vcard
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Re: [Asterisk-Users] How to remove standard ISDN drivers from RedHat
On Mon, Aug 15, 2005 at 10:23:25PM +0200, Remco Barende wrote: I have newly installed a RedHat 4.0 EL rebuild. The install was done without the ISDN card present. After disabling kudzu and haldaemon I inserted the card. Stil that *($^%$($^!! kudzu shit modified my config and is loading hisax, crc_ccit and isn modules. Even worse, they do not appear in /etc/modprobe.conf which means that that f*cking kudzu added the modules to initrd. Huh? BTW: why won't you disable kudzu? What happens if you re-create the initrd? I have googled for hours and browsed through all the redhat docs but I cannot find how to remove these modules. All the docs mention is to 'simply comment them out from modprobe.conf' well, they aren't there. Does anybody know how I can remove these modules? They are really a pain in the ass because now I cannot start asterisk from the init scripts only from rc.local Sure you can. As a last resort, manually rmmod and only then modprobe, or use two different init.d scripts. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff-0.2.0-RC8n problems and kernel panic
Hi list! On a newly installed RHEL 4 box I'm trying to install bristuff-0.2.0-RC8n. Everything did compile but I am running into some problems with the zaphfc driver. First of all when I load zaphfc *before* zaptel (yes I know I shouldn't do that) I get a kernel panic and the box hangs. Not so nice, especially when you are trying to fix stuff from remote locations. But ok. Now for the real trouble, when I do make load in zaphfc I get this: make -C /usr/src/linux-2.6 SUBDIRS=/tmp/bristuff-0.2.0-RC8n/zaphfc ZAP=-I/tmp/bristuff-0.2.0-RC8n/zaptel-1.0.9 modules make[1]: Entering directory `/usr/src/kernels/2.6.9-11.EL-x86_64' Building modules, stage 2. MODPOST *** Warning: zt_register [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! *** Warning: zt_receive [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! *** Warning: zt_transmit [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! *** Warning: zt_ec_chunk [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! *** Warning: zt_unregister [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! make[1]: Leaving directory `/usr/src/kernels/2.6.9-11.EL-x86_64' modprobe zaptel insmod ./zaphfc.ko ztcfg -v Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) 3 channels configured. Notice: Configuration file is /etc/zaptel.conf line 8: Unable to open master device '/dev/zap/ctl' I guess (hope) the warnings are nothing serious but the message about /dev/zap/ctl is. (I did read README.udev and added the lines.) Rebooting the box didn't help. And when I try to start asterisk: Aug 15 23:25:51 WARNING[6454]: chan_zap.c:933 zt_open: Unable to specify channel 1: No such device or address Aug 15 23:25:51 ERROR[6454]: chan_zap.c:6484 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Aug 15 23:25:51 ERROR[6454]: chan_zap.c:10329 setup_zap: Unable to register channel '1-2' Aug 15 23:25:51 WARNING[6454]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Aug 15 23:25:51 WARNING[6454]: loader.c:440 load_modules: Loading module chan_zap.so failed! Ideas anyone? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 8 FXS in Asterisk Server
Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI cards supporting 8 FXS interfaces, but without success. does anyone know such hardware? Thanks in advance, Roland ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 8 FXS in Asterisk Server
On Mon, 2005-08-15 at 23:32 +0200, Roland Zagler wrote: Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI cards supporting 8 FXS interfaces, but without success. does anyone know such hardware? Thanks in advance, Roland Either a T1 card to channel bank with 8 FXS channels, which is expensive but allows for great expandability down the line, or you'll have to look at ATAs/gateways (networking in other words). If the computer only has 1 PCI slot, how would you expect to fit 2 brackets worth of FXS connectors if you went with an internal solution? (I'm thinking 1 PCI slot, 1 backside bracket.) I know there is a Zaptel USB module, but don't know of any hardware or compatibility information. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo calibration with ztmonitor and a test line from a telco
I have been doing a bit of this too lately. This was also useful. http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html Dan What about for PRI lines? We get echo every now and then. The docs link above references FXO lines. We have none. But we do have 4 PRIs. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dnsmgr
Hello, What's dnsmgr ? Anybody could tell mr more? cat /etc/asterisk/dnsmgr.conf [general] ;enable=yes ; enable creation of managed DNS lookups ; default is 'no' ;refreshinterval=1200 ; refresh managed DNS lookups every n seconds ; default is 300 (5 minutes)serveur1:~# Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
This service has been working well lately, but as of this morning is promptly blowing off IAX connections with the dreaded 'No Authority Found' error. Any concrete info greatly appreciated! Dr G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
As of 22:45 GMT it's working for me Jerry Glomph Black wrote: This service has been working well lately, but as of this morning is promptly blowing off IAX connections with the dreaded 'No Authority Found' error. Any concrete info greatly appreciated! Dr G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.7/70 - Release Date: 11/08/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco IP Phone- 7905G
Joseph: Thank you for the help. Orlando From: Joseph [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco IP Phone- 7905G Date: Sat, 13 Aug 2005 11:56:51 -0400 Orlando Guitián wrote: Has anybody used a Cisco 7905G or similar model with Asterisk using skinny? How can i set it up with an asterisk box? Are you using the latest version of chan_sccp? http://www.voip-info.org/tiki-index.php?page=chan_sccp2 The driver link can be gotten directly from there. There is a mailing list just for chan_sccp. Try the latest version and let us know what happens when you try to register. -- respectfully, Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 8 FXS in Asterisk Server
Easy and cheap. Get two gateways AG-468 (each have 4 FXS ports) made by Atcom http://www.voip-info.org/tiki-index.php?page=Atcom one is about 88/ea I have two on the way and will let you know how it works. -- #Joseph On Mon, 2005-08-15 at 23:32 +0200, Roland Zagler wrote: Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI cards supporting 8 FXS interfaces, but without success. does anyone know such hardware? Thanks in advance, Roland ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 8 FXS in Asterisk Server
Thanks for the hint, where have you bought them? Roland -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Tuesday, August 16, 2005 12:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 8 FXS in Asterisk Server Easy and cheap. Get two gateways AG-468 (each have 4 FXS ports) made by Atcom http://www.voip-info.org/tiki-index.php?page=Atcom one is about 88/ea I have two on the way and will let you know how it works. -- #Joseph On Mon, 2005-08-15 at 23:32 +0200, Roland Zagler wrote: Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI cards supporting 8 FXS interfaces, but without success. does anyone know such hardware? Thanks in advance, Roland ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NAT'd Snom360 problems
Here is my setup: * is on a NAT'd subnet, but also has an externally routable IP address. I have a Snom360 that's external to this and behind NAT. The Snom360 can call other phones in * subnet (by their internal extension numbers) and voice is transmitted fine; however, when I attempt to check voicemail (or any * voice recordings for that matter) I can't hear them. The phone just connects to voicemail for 4-5 seconds and then disconnects. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuration to get CallerID working in New Zealand
Hi All. We have 2 clone x100ps and they work well but we cant get callerID working, they should work right out of the box so if anyone in NZ has a working callerID setup if they could send me the Zapata.conf config that would be great Cheers Tristram ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
For me to - Original Message - From: Mat Stace, Colewood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 15, 2005 5:46 PM Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? As of 22:45 GMT it's working for me Jerry Glomph Black wrote: This service has been working well lately, but as of this morning is promptly blowing off IAX connections with the dreaded 'No Authority Found' error. Any concrete info greatly appreciated! Dr G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.7/70 - Release Date: 11/08/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax Issues
On Mon, 2005-08-15 at 12:40 -0700, Matt wrote: I have a user who has a fax machine plugged into an ATA. They are able to SEND faxes just fine. Faxes go through wonderfully. However, when someone tries to send them a fax, their fax machine never receives it. And eventually the sending machine just errors out. Any thoughts? Need more info! Is the fax plugged to dedicated port on ATA. What kind of ATA is it? Use NVBackgroundDetect works OK. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 8 FXS in Asterisk Server
Joseph wrote: Easy and cheap. Get two gateways AG-468 (each have 4 FXS ports) made by Atcom http://www.voip-info.org/tiki-index.php?page=Atcom one is about 88/ea I have two on the way and will let you know how it works. I would be interested in knowing how these work as well Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie FreeWorldDial 689482 smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Simple Fax question
Strange things. When I run the RxFAX command through an internally dialed extension, I can *hear* fax tones, meaning, I presume, that the RxFAX application is running. In fact, doing a show application confirms that. So, I'm presuming RxFAX application is talking as it should. However, inbound fax calls (tones) are not being detected. I know that my extensions file is corect. I am ONLY running SIP. I've seen this question posted here several times and no one answers: is it possible to have fax tones detected if you are only running SIP protocol without any digium hardware or cards? Not running IAX protocol. Can some kind soul please answer this simple question, if known. Running Asterisk CVS-HEAD-05/23/05 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
