Re: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
Jamie Carl wrote: Bob Knight wrote: There is a linux package called mbrowse that you can use with your mediatrix mibs. I can get and walk everything in my 1204's. For some reason I have not had any success with writes, but I have not spent that much time on it. I don't even have the MIBs which is half the problem. I can do certain things using windoze SNMP software, but not exactly being a guru on SNMP i'm guessing that without the MIBs i'm pretty much stuffed. Anyone with MIBs they can send me? hehe Please? :) I have MIBs for whatever version I am running that I am more than happy to share. Anyone know where I can place these for public access. Sort of like the freedomphones site for Polycom. We could then put pointers on the wiki. Thanks for the info tho. If mbrowse is console based it will be very useful. :) It has gui (X, gtk I think) if that is what you mean by console based. I can ssh into a remote * server and do get walks on my 1204's. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual.
Gonzalo, I have an APA III-4FXO and I tried using your configurations, I received the message below: -- Executing Dial(SIP/2010-edfc, SIP/[EMAIL PROTECTED]) in new stack Sep 6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814bf0c (len 774) to 192.168.199.5 returned -1: Operation not permitted -- Called [EMAIL PROTECTED] Sep 6 16:54:54 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814409c (len 360) to 192.168.199.5 returned -1: Operation not permitted == Spawn extension (from-sip, 92217008, 1) exited non-zero on 'SIP/2010-edfc' Sep 6 16:54:56 WARNING[1125350192]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814409c (len 360) to 192.168.199.5 returned -1: Operation not permitted Sep 6 16:54:57 WARNING[1125350192]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) Sep 6 16:54:59 WARNING[1125350192]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814409c (len 360) to 192.168.199.5 returned -1: Operation not permitted Sep 6 16:55:00 WARNING[1125350192]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) My configurations are: SIP.Mib sipUAServerStaticRegistrarHost = 192.168.199.4 sipUAServerStaticRegistrarPort = 5060 sipUAServerStaticProxyHost = 192.168.199.4 sipUAServerStaticProxyPort = 5060 sipUAServerStaticOutboundProxyHost = 192.168.199.4 sipUAServerStaticOutboundProxyPort = 5060 sipUA1PrefixCCAndAC = 0 sipUA1MainAlias = sipUA1FriendlyName = sipUA1OtherAliases = sipUA1MustUseSessionTimers = 0 sipUA1MaximumSessionExpirationDelay = 60 sipUA1AuthUsrPwd = sipUA1AuthValid = 1 My sip.conf and extension.conf are the same that you send. Any help will be appreciated. Kind regards, Miguel Date: Sat, 4 Sep 2004 15:38:42 -0700 (PDT) From: Gonzalo Gasca Meza [EMAIL PROTECTED] Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Here is my configuration for MEdiatrix 1204, by default the 1204 strips one digit, so it is not necessary to use: To dial OUTSIDE EXTENSIONS.CONF [locales] ;ignorepat = 9 exten = _9,1,Dial(SIP/[EMAIL PROTECTED]) exten = _9,2,Congestion exten = _9,102,Congestion To receive calls [from-pstn] ;Incoming calls from Mediatrix 1204, the 1204, sends an invite to [EMAIL PROTECTED] exten = ,1,Dial(SIP/100,20) exten = ,2,Voicemail(u100) exten = ,102,Voicemail(b100) exten = ,103,Hangup *** SIP.CONF ;Mediatrix Telecomm 1204 [Mediatrix] type=peer host=110.10.200.10 mask=255.255.255.255 context=from-sip qualify=yes canreinvite=yes disallow=g729 nat = yes In MEdiatrix 1204 use a program called Unit Manager Network a Configure the first port as extension for port 1, in option SIP. as user agent. also edit registar an dproxy SIP as the IP address of Asterisk. Works VERY GOOD with one line, although i have seen some scenarios with more than 1 line which experince problems. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
Bob Knight wrote: I have MIBs for whatever version I am running that I am more than happy to share. Anyone know where I can place these for public access. Sort of like the freedomphones site for Polycom. We could then put pointers on the wiki. Thanks for the info tho. If mbrowse is console based it will be very useful. :) It has gui (X, gtk I think) if that is what you mean by console based. I can ssh into a remote * server and do get walks on my 1204's. Bob, I've managed to source the MIBs from another extremely helpful list member so hopefully I'm all sorted. :) As for posting them, as I'm sure there are others out there that are interested, there is a website called www.mibdepot.com which is trying to collect as many MIBs as possible and currently has a request for the APA III-4FXO MIB. If you email it to the webmaster of that site he'll post it as part of his collection. I found this site while I was looking for it myself so hopefully others will look there too as they already have quite a few MIBs available. Jamie ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
Thanks to everyone for their help and comments on this. You've all been very helpful. I've actually got outbound calls working on it fine right now without having to change the configuration on the Mediatrix box at all, as I don't have the Unit Manager Software at the moment. Outbount seems to work well but without inbound it means I can't put it in place for general use. I have my 'reseller' tracking down the software for me right now so hopefully he'll be able to find it for me. :) Asterisk doesn't seem to have any issues working with the APA III-4FXO at all as yet. Thanks again guys. J Gonzalo Gasca Meza wrote: Here is my configuration for MEdiatrix 1204, by default the 1204 strips one digit, so it is not necessary to use: To dial OUTSIDE EXTENSIONS.CONF [locales] ;ignorepat = 9 exten = _9,1,Dial(SIP/[EMAIL PROTECTED] mailto:SIP/[EMAIL PROTECTED]) exten = _9,2,Congestion exten = _9,102,Congestion To receive calls [from-pstn] ;Incoming calls from Mediatrix 1204, the 1204, sends an invite to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] exten = ,1,Dial(SIP/100,20) exten = ,2,Voicemail(u100) exten = ,102,Voicemail(b100) exten = ,103,Hangup *** SIP.CONF ;Mediatrix Telecomm 1204 [Mediatrix] type=peer host=110.10.200.10 mask=255.255.255.255 context=from-sip qualify=yes canreinvite=yes disallow=g729 nat = yes In MEdiatrix 1204 use a program called Unit Manager Network a Configure the first port as extension for port 1, in option SIP. as user agent. also edit registar an dproxy SIP as the IP address of Asterisk. Works VERY GOOD with one line, although i have seen some scenarios with more than 1 line which experince problems. Do you Yahoo!? Win 1 of 4,000 free domain names from Yahoo! Enter now http://us.rd.yahoo.com/evt=26640/*http://promotions.yahoo.com/goldrush. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
Jamie Carl wrote: Thanks to everyone for their help and comments on this. You've all been very helpful. I've actually got outbound calls working on it fine right now without having to change the configuration on the Mediatrix box at all, as I don't have the Unit Manager Software at the moment. Outbount seems to work well but without inbound it means I can't put it in place for general use. I have my 'reseller' tracking down the software for me right now so hopefully he'll be able to find it for me. :) Asterisk doesn't seem to have any issues working with the APA III-4FXO at all as yet. Thanks again guys. There is a linux package called mbrowse that you can use with your mediatrix mibs. I can get and walk everything in my 1204's. For some reason I have not had any success with writes, but I have not spent that much time on it. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
Bob Knight wrote: There is a linux package called mbrowse that you can use with your mediatrix mibs. I can get and walk everything in my 1204's. For some reason I have not had any success with writes, but I have not spent that much time on it. I don't even have the MIBs which is half the problem. I can do certain things using windoze SNMP software, but not exactly being a guru on SNMP i'm guessing that without the MIBs i'm pretty much stuffed. Anyone with MIBs they can send me? hehe Please? :) Thanks for the info tho. If mbrowse is console based it will be very useful. :) J ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
Here is my configuration for MEdiatrix 1204, by default the 1204 strips one digit, so it is not necessary to use: To dial OUTSIDE EXTENSIONS.CONF [locales];ignorepat = 9 exten = _9,1,Dial(SIP/[EMAIL PROTECTED])exten = _9,2,Congestionexten = _9,102,Congestion To receive calls [from-pstn];Incoming calls from Mediatrix 1204, the 1204, sends an invite to [EMAIL PROTECTED] exten = ,1,Dial(SIP/100,20)exten = ,2,Voicemail(u100)exten = ,102,Voicemail(b100)exten = ,103,Hangup *** SIP.CONF ;Mediatrix Telecomm 1204[Mediatrix]type=peerhost=110.10.200.10mask=255.255.255.255context=from-sipqualify=yescanreinvite=yesdisallow=g729nat = yes In MEdiatrix 1204 use a program called Unit Manager Network a Configure the first port as extension for port 1, in option SIP. as user agent. also edit registar an dproxy SIP as the IP address of Asterisk. Works VERY GOOD with one line, although i have seen some scenarios with more than 1 line which experince problems. Do you Yahoo!? Win 1 of 4,000 free domain names from Yahoo! Enter now.___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual?
Hi all, I just picked myself up a Mediatrix FXO SIP gateway to play around with and hook into Asterisk but have no documentation. Are there default passwords or IP's that I need to know if I do a factory reset? Or better still, would anyone have a User Manual they could send my way? Any help would be appreciated. TIA. Jamie ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual?
I just picked myself up a Mediatrix FXO SIP gateway to play around with and hook into Asterisk but have no documentation. I spent a substantial amount of time evaluating the 1204 box back in the January timeframe, and then returned it to the reseller. I can answer some of your questions but not all. The Mediatrix products are not bad at all, but they can only be configured via a Windows SNMP application that comes with each firmware version on the 1204. There is no telnet or web interface. Without that app, getting the box to work with asterisk will not be possible. Mediatrix does not have any direct support; they expect their resellers to support the user, and they expect the reseller to invoice you for each software upgrade, etc. The box is shipped from Mediatrix with both H.323 and SIP software, however the reseller is suppose to only give you one or the other. (There are different model numbers for those two, but its the same box, just a different software load.) The software required to configure the box _must_ match the firware running in the box. When I was testing, they were at v1.4.6.20, and each firmware release required a deinstall and reinstall of the configuration software. I tried two or three different SIP firmware versions to address different problems, and had to go through the process multiple times. The firmware upgrade process actually forces you to start the process with the old configuration software (on Windows), initiate the upgrade, and sometime prior to rebooting the 1204, deinstall and reinstall the new configuration software so you can interact with the new firmware. Its a real pain. Given where you're at with the box, you'll probably need to get the latest sip firmware, the manual that goes with that version, and the configuration software that matches that firmware. Since they rely on the use of SNMP to configure the box, you'll spend a fair amount of time working with the MIBs within the configuration software trying to find the parameters necessary to accomplish some task. The admin manual is pretty good, but finding the words (and appropriate MIB variable) to match an asterisk function is far less then ideal. (The more you know about SNMP, the easier it is.) Are there default passwords or IP's that I need to know if I do a factory reset? A factory reset will but the box into dhcp mode, and will obtain an IP address on subsequent reboots. The SNMP community string (password) defaults to public, and in January 2004, could not be changed to anything else period. Again, without their SNMP configuration software you'll not be able to get the box configured properly. Or better still, would anyone have a User Manual they could send my way? Any help would be appreciated. Mediatrix still seems to be focused on the toll bypass business, and intended the 1204 (fxo) to be used in conjunction with the 1104 (fxs) box. As a result, there are a fair number of non-sip-compliant protocol 'enhancements' in their firmware, however the box can be made to work with *. There are a few users on this list that are using the 1204 successfully. The box does some strange things that made it unusable for me. Like it detects ring cadance on the first incoming call following a reboot and applies that same cadance to all four lines. In my case, I had one pstn line (of four) with a different cadance which caused the box to never answer incoming calls on that port. :( There's also no nice way to pick a specific pstn port number when making outgoing calls via the box. You'll need to muck around with setting a 'callerid' in *, and then set a matching parameter within the 1204 to recognize that callerid on a per-port basis. The box will then use that port for the call. It's default config is to use 'silence suppression' which will cause very choppy sound with asterisk, so that config parameter will need to change as well. To get the box to work (and be legal), you'll need to contact a reseller and order the current software from them. That cdrom will include the user manual (*.pdf), the configuration software (for Windows only), and the binary image needed to upgrade it. (Be sure to specify either H.323 or SIP as they won't ship both.) You'll also need a tftp server to complete the process. And, be constantly aware that if you discover what you believe to be a firmware problem, they will want to charge you again for the next version. There are lots of different reasons for not using that box in a production business environment (mostly revolving around support, enhancements, bug fixes, cost of ownership, potential bankrupcy again) but for the home or small office it functions rather well. Good luck... Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
