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fredtheman

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Everything posted by fredtheman

  1. Interesting. I could write an app that checks for changes to this file every 30 seconds, upload it to a web server if it changed where it will be parsed by a server-side script to update a web page. Thanks for the pointer
  2. Hi Does someone know of a way for Axon to log calls and send this information to remote hosts so that everyone in the company can see who called today, regardless of their having an IP phone and being in a group extension? The idea would be that, whenever a call comes in (usually from the PSTN line), Axon... - either notifies a web server through a URL (eg. http://www.acme.com/calls.php?name=JohnDoe&number=123 where name and number are the caller ID information as sent by the SIP gateway to Axon), and the server-side script can update a web page that lists the calls for the day - or sends a message to all the hosts that have a desktop client that pops up every time a call comes in and also serves as a call logger. Any idea if Axon can be made to do something like that? Thank you.
  3. For those struggling with getting the 3102 to notify a PBX of an incoming call, read this: http://wiki.pbxnsip.com/wiki/index.php/Lin...PA-3102_FXS/FXO This solved both the notification issue and the missing sound in one or both directions. Works like a charm now.
  4. Didn't you answer it yourself above? This is also where you tell Axon to NOT go off-hook and play music on hold, and instead, let the incoming call RING the extension of extension group. Hold Options while waiting > Ringing Only
  5. Outstanding issues left - When the softphone goes off-hook, I can hear the remote PSTN caller, but the remote caller can't hear me through the headset microphone (Skype works fine, so it's not the hardware or software part of the sound card.) => I still can't figure that one out. There's no firewall between the Linksys and the PC where Axon and X-Free are running, both are connected on the same switch through the Ethernet port. There are so many settings regarding ports, RTP, etc., that I don't know where to look, what to try. - When a call comes in, the Linksys goes off-hook immediately, and then calls the softphone: As a result, even if the softphone never takes the call, the caller will be charged for the call. I didn't find how to change this. By default, Axon intends to play music on-hold. Either because I didn't install IVM or because of the issue with sound above, I got silence. Changing its default to RING solved the issue: Axon tells the Linksys to go off-hook only when the SIP client actually answers the call. Thanks for any hint
  6. Great news! The 3102 finally notifies Axon of incoming calls The trick is to ignore the documents above (either they're specific to the 3000 or they no longer apply to the latest firmware, 3.2.10), especially the part about dial plans. From what I've seen, I get the impression that the Linksys was built, foremost, as an ATA, not as an SIP gateway. This would explain why incoming calls are first sent to the FXS, and will only be forwarded to Axon if setting it up that way in the User 1 section. For those interested, here's what needs to be done in Axon and the 3102 to handle incoming FXO calls: AXON Launch Axon, and aim your browser to its web server (http://localhost:81). Create an extension for the FXS device (eg. fxs/fxs) Modify the default external line to create an account for the Linksys to use to register with Axon, eg. fxo/fxo. Don't put the Linksys IP address in Server: Either Axon 1.08 doesn't support this yet, or the relevant info must be added somewhere in teh Linksys but I haven't looked into this yet as I don't need the Linksys to make outgoing calls. In Extensions, click on "Setup Details" for extension 101, and write down the password. You'll need it to set up the X-Lite free softphone next. X-Free softphone Next, download and install this application. Register with Axon using 101 as UserID, and the password for this extension. The softphone will register with Axon and be ready to receive calls. Linksys Finally, aim your browser to the Linksys embedded web server, log on as admin/advanced, and change the following parts in the Voice section: Line 1 "Line Enable" = yes (or the 3102 won't notify Axon) "Proxy and Registration" : Proxy = IP address of Axon "Register" = yes or no, it worked either way (but I guess it should be yes for the FXS to be reachable) "Subscriber Information" : "User ID" and "Password" must match an extension in Axon PSTN Line Proxy and Registration : Proxy = IP of Axon, Register = yes Subscriber Information : User ID and password must match the information entered in Axon for the default external line PSTN-To-VoIP Gateway Setup : "PSTN Ring Thru Line 1" = YES (important!), "PSTN CID For VoIP CID" = yes (important! otherwise, caller ID from the phone company won't be passed on to the IP phone) User 1 Call Forward Settings: Cfwd All Dest=ID that matches the one that you set up for the default external line in Axon (it means that all incoming calls will be forwarded from the FXS to the FXO) NB: By default, the FXS ("Line 1") uses UDP 5060 and the FXO ("PSTN Line") uses UDP 5070. I read here that Axon expects the FXO to use 5060 to be usable for outgoing calls. As I only need the Linksys to handle incoming calls from the PSTN line, I haven't tested this yet. Now, call into the Linksys from a remote phone: The Linksys should detect the call, see that all incoming calls should be forwarded from the FXS to the Axon server, which should ring extension group 701 that includes 101 and 102. Since the softphone is registered as extension 101, you should get a ring. Outstanding issues - When the softphone goes off-hook, I can hear the remote caller, but the remote caller can't hear me (yes, the cables are OK, and I can see the MIC volume going way up in X-Free when I talk into it): Is it due to some RTP business? Incidently, how do two SIP devices (here, the Linksys and the X-Free softphone) know which UDP ports to use with RTP? They send this information during the SIP call setup? - When a call comes in, the Linksys goes off-hook immediately, and then calls the softphone: As a result, even if the softphone never takes the call, the caller will be charged for the call. I didn't find how to change this. Any help appreciated for the last two items Fred.
  7. Axon still doesn't display any message when I call the Linksys 3102 FXO from my cellphone, but at least, the Linksys does send some information to a syslog server (cellphone number partially obscured to protect the innocents): Aug 16 07:00:59 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"FXO:Start CNDD " Aug 16 07:00:59 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"FXO:CNDD name=, number=062805XXXX " Aug 16 07:00:59 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"FXO:Stop CNDD " Aug 16 07:00:59 Romance Daylight Time,,192.168.0.1,local3,DEBUG,,"FXO:CNDD Name= Phone=062805XXXX " -------- HUNG UP CELLPHONE----------------- Aug 16 07:01:15 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"AUD:Stop PSTN Tone " Aug 16 07:01:15 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"AUD:Stop PSTN Tone " Aug 16 07:01:21 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"AUD:Stop PSTN Tone " Aug 16 07:01:21 Romance Daylight Time,,192.168.0.1,local3,DEBUG,,"FXO:On Hook " Aug 16 07:01:21 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"AUD:Stop PSTN Tone " Aug 16 07:01:21 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"FXO:Stop CNDD " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"CC:Clean Up " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"--- OBJ POOL STAT --- " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:RTPRXB = 96 ( 96 192) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:RTPREB = 40 ( 40 48) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:RTPTXB = 64 ( 64 108) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:TIMEOU = 110 (120 40) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:SIPCOR = 0 ( 1 28) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:SIPCTS = 32 ( 32 568) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:SIPSTS = 32 ( 32 3492) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:SIPAUS = 5 ( 8 588) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:SIPDLG = 10 ( 10 148) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:SIPSES = 12 ( 12 7936) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:SIPREG = 2 ( 4 292) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:SIPLIN = 0 ( 2 140) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:ÿ" Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,,"OP:STUNTS = 16 ( 16 68) " Aug 16 07:01:37 Romance Daylight Time,,192.168.0.1,local2,DEBUG,," " FWIW, I followed the instructions in the following documents... FXO Adapters Installation Guide Sipura 3000 Configuration External Line to SPA3000 DONT WORK, SPA3000 configuration problem AXON with SPA-3102, SPA-3102 is the new SPA-3000 + router ... and uploaded a copy of the Linksys configuration: http://codecomplete.free.fr/linksys.html Me thinks it's really something basic in the Linksys configuration, but any help much appreciated :-) Thx Fred. PS : Here's what I did (Axon = 192.168.0.2, Linksys = 192.168.0.1) 1. In Axon, in Extensions, create an account fxo/fxo so the Linksys can register its FXO port (I don't care about the FXS at this point) 2. Still in Axon, in External line, modified the Default item, Server = 192.168.0.1, ID = fxo, Password = fxo, Group = 701). This triggers the familiar "Attempting to register sip:fxo@192.168.0.1, Register attempt for sip:fxo@192.168.0.1 failed, 501 Not Implemented) that others have seen. Using or not using a password makes no difference. I don't see how Axon could possibly use the FXO to make outgoing calls without knowing the IP address of the Linksys, though... 3. In Linksys, in "PSTN Line", changed the FXO port from 5061 to 5060 (and FXS from 5060 to 5070), filled the "Proxy and Registration" and "Subscriber Information" with relevant information : The Linksys successfully registers with Axon. Also modified the "Dial Plan 1" with "(S0<:701@192.168.0.2:5060>)" (w/out the quotes).
  8. fredtheman

    SPA 3000 Help

    Same here. I can see the FXS plug and a softphone registering with Axon, but nothing happens when a call comes in through the FXO port, apart from the FXS phone ringing (but since it's not part of the ringing group, I assume it's just a default setting in the Linksys 3102 I'm using). So far, I haven't found any documentation on how to set it up. Even after following the FXO Setup Guide (ie. in the PSTN Line section, "Dial Plans" sub-section, setting "Dial Plan 1" to eg. "(S0<:701@192.168.0.2:5060>) so that incoming calls would trigger ring group 701 (extensions 101, followed by 102))... but no message shows up in the log :-( I keep staring at all those options in the Voice menu...
  9. Hello The 3102 just got here, I plugged it into the hub, fired a web browser... and I'm stuck :-) Can a kind soul who has successfully configured this SIP gateway for use with Axon tell me how to get started? FWIW, I will only use the Phone and LAN connections, as I already have an Internet router. The Linksys is just there to handle PSTN incoming and outgoing connections Thank you Fred.
  10. Posts here aren't getting much reply from NCH anyway, but for sales questions... http://nch.invisionzone.com/index.php?showtopic=2051
  11. Hello I installed 1.07 and read the How to route/divert/transfer a call to an external phone number section. Before I spend more time learning Axon, I'd like to make sure that this means that I can buy an SIP gateway to connect to the PSTN, and have both local and remote IP phones register to Axon to receive incoming calls. In other words, when a call comes in through the SIP gateway (eg. Sipura/Linksys FXO to SIP thingie), I want Axon to not only ring local phones but also ring some remote IP phones without answering the call, ie. I don't want users to have to type any extension before diving into a dial plan. The goal is to be able to work from two different offices, and have anyone answer incoming calls. That way, customers can't tell where we are. Can Axon 1.07 do this? Thank you Fred.
  12. fredtheman

    AXON with SPA-3102

    As I'm sure a lot of users are thinking of using the SPA 3xxx to work with Axon, it might be useful to explain how you got it to work, and whether you would recommend it :-)
  13. There seems to be a lot of information on how to get the Sipura to pass on CID information unchanged to handsets/softphones: http://www.google.com/search?q=PSTN+Caller+ID+Pattern Please report back, as I'm about to order a Sipura 3xxx :-)
  14. ... and if that friend has successfully used Axon in France, I'm very interested in his recommendations on what SIP device to buy. Thanks Fred.
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