**Error: Can't convert VSP into Fixnum

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qubo
Posts: 127
Joined: Tue Apr 12, 2011 6:11 am

Re: **Error: Can't convert VSP into Fixnum

Post by qubo » Fri May 13, 2011 5:38 am

thanks so much Mike I put the code you gave me and made the sip:gvny.inquss@sipsorcery.com but the USA numbers that are incoming from my google voice put the code 52. I saw the code and the console does not show gvny.inquss redirect maybe the server take some time to update. I think it was fast so I will test tomorrow.


bye the way Mike How I can make all incoming calls to sound in all my devices at same time ??? I used to have one incoming dial plan that make it sound all my extensions at same time this was the code

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sys.Dial("inquss@local&IPOD@local&portech@local")
so to take advantage of your GREAT DIALPLAN I want to put your DIAL PlAN for outgoing and incoming but I want still all my extensions sound and answer the first availabe. so Ho I can put this code in your dial plan ???



Other question. How I Can make that each DID show the prefix in the phone display to see what DID the call arrived from ?

example

A for Nextel DID would forward to A.inquss@sipsorcery
B for RealE DID would forward to B.inquss@sipsorcery
C for Casa DID would forward to C.inquss@sipsorcery
D for DF DID would forward to D.inquss@sipsorcery
M for Monterrey DID would forward to M.inquss@sipsorcery
V for Voxalot DID would forward to V.inquss@sipsorcery

I know I need to chage every DID but I want to know how this config would be set in MAIN area

so the calls for mexico could be A +52 33 xxxx so I know this call got into the nextel DID
or C +52 33xxx I know the incoming call is target Casa DID

thanks so much I will tell you tomorrow the test of the gvny.inquss@sipsorcery.com to make if the incoming calls show as +1 for US. since the DID show its already set the gvny but the console didnt show that

thanks for all I APPRECIATE SO MUCH WHAT YOU DID AND DO..

MikeTelis
Posts: 1582
Joined: Wed Jul 30, 2008 6:48 am

Re: **Error: Can't convert VSP into Fixnum

Post by MikeTelis » Fri May 13, 2011 7:49 am

qubo wrote:thanks so much Mike I put the code you gave me and made the sip:gvny.inquss@sipsorcery.com but the USA numbers that are incoming from my google voice put the code 52. I saw the code and the console does not show gvny.inquss redirect maybe the server take some time to update. I think it was fast so I will test tomorrow.
I need to see the Console log for such a call. Of course, I didn't test the code and it might be bugged.
qubo wrote: How I can make all incoming calls to sound in all my devices at same time ?
Just make all devices register to the same SIP account (inquss@sipsorcery.com). Sipsorcery server will forward incoming calls to all bound devices.

Of course, you can multi-forward ("spread") the calls manually like you did before but it's only necessary when you need to handle incoming calls in more sophisticated manner. For example, you may want to ring your ATA immediately and delay ringing to iPod (make it ring in 10 sec if the call wasn't answered on ATA).
How I Can make that each DID show the prefix in the phone display to see what DID the call arrived from ?
You could add @trunk to "display name" or "user" portion of the SIP URI. It depends on your SIP devices, they all display this information differently. Some show only display name, the others should both display name and user. Of course, the length of each portion is also limited (by screen width). You need to do it yourself, by try and error. Please refer to the Sipsorcery help on dialplan methods; check out sys.SetFromHeader() method and [fu], [fd] modifiers of sys.Dial() method. You'll need to modify the incomingCall() method in your dialplan, refer to the dialplan manual, @cname, @cid, @trunk.

qubo
Posts: 127
Joined: Tue Apr 12, 2011 6:11 am

Re: **Error: Can't convert VSP into Fixnum

Post by qubo » Fri May 13, 2011 6:17 pm

HI. Mike

I saw that I can set @trunk

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case @trunk
  when "C" then sys.Dial("#{@user}@local")    # Casa Trunk
  when "N" then sys.Dial("#{@user}@local")    # Nextel Trunk
  when "R" then sys.Dial("#{@user}@local")    # RealE
 when "D" then sys.Dial("#{@user}@local")    # DF Trunk
 when "M" then sys.Dial("#{@user}@local")    # Monterrey Trunk
 when "I" then sys.Dial("#{@user}@local")    # IdeaL TRunk
end
this is the code I need. but as we have the +52 Code to identify if it mexican call or USA call so I was wondering if the rule you create for country code could add the trunk that is only 1 letter to save display width.
example and incoming call to my trunk Nextel would show as N52 1234567890 and a Call to House would be C52 1234567890 this way I know that it is a Mexican call and know the trunk that is arriving the call

your code actually is this

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if sys.In               # If incoming call...
       prs = req.URI.User.split('.')  # parse User into chunks
    @trunk = prs[-2]               # get trunk name
    @user  = prs[-1]               # called user name

    @cid = req.Header.from.FromURI.User.to_s    # Get caller ID

    case @trunk.to_s.downcase
      when 'gvny', 'gvsm'   # called at Google Voice DID
        @cid = ('1' + @cid) if @cid =~ /^[2-9]\d\d[2-9]\d{6}$/  # Prepend 10-digit numbers with "1" (US country code)
        @cid.sub!(/^(\+|00|011)/,'')                            #remove int'l prefix (if present)
      else                  # Mexican DID
        @cid = (Country + @cid) if @cid =~ /^[2-9]\d{9}$/ # Prepend numbers with "52" (Mexico country code)
    end

    # Check CNAM first. If not found and US number, try to lookup caller's name in Whitepages

    if !(@cname = keys_to_ENUM(CNAM)[@cid]) && @cid =~ /^1([2-9]\d\d[2-9]\d{6})$/ && defined?(WP_key)
      url = "http://query.yahooapis.com/v1/public/yql?q=select%20*%20from%20xml%20where%20url%3D'http%3A%2F%2Fapi.whitepages.com%2Freverse_phone%2F1.0%2F%3Fphone%3D#{$1}%3Bapi_key%3D#{WP_key}'%20and%20itemPath%3D'wp.listings.listing'&format=json"
      if js = sys.WebGet(url,4).to_s
        @cname, dname, city, state = %w(businessname displayname city state).map {|x| js =~ /"#{x}":"([^"]+)"/; $1}
        @cname ||= dname; @cname ||= "#{city}, #{state}" if city && state
      end
    end

    sys.Log("Caller's number: '#{@cid}'"); sys.Log("Caller's name:   '#{@cname}'") if @cname
    incomingCall()        # forward incoming call

By The way the code for the test to gvny.inquss that is still showing the code 52 instead of +1 is this NOTE: the call was made for other mexican line and dial to my google voice abroad, the incoming call for all gv are ipcomms trunk that is config to forward calls to gvny.inquss@sipsorcery.com but in the console log is not any trace that the call is comming from gvny.inquss@sipsorcery.com and dont know why. since the trunk ipcomm is set correctly

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:00:834 sip1(16328): ** Call from "3315920xxx" <sip:3315920766@64.154.41.150>;tag=as1f3d37f8 to inquss **
DialPlan 18:23:00:850 sip1(16328): Local time: 05/13/2011 13:23
DialPlan 18:23:00:866 sip1(16328): Caller's number: '523315920xxx'
DialPlan 18:23:00:912 sip1(16328): URI dialing: inquss@local[fu=523315920xxx]
DialPlan 18:23:00:912 sip1(16328): Commencing Dial with: inquba@local[fu=523315920xxx].
DialPlan 18:23:00:928 sip1(16328): Call leg is for local domain looking up bindings for inquss@sipsorcery.com for call leg inquss@local.
DialPlan 18:23:00:944 sip1(16328): 3 found for inquss@sipsorcery.com.
DialPlan 18:23:00:944 sip1(16328): ForkCall commencing call leg to sip:inquss@187.133.147.183:49503;rinstance=E09D708A;transport=tcp.
DialPlan 18:23:00:944 sip1(16328): ForkCall commencing call leg to sip:inquss@187.133.147.183:65246.
DialPlan 18:23:00:944 sip1(16328): ForkCall commencing call leg to sip:inquss@187.133.147.183:65176.
DialPlan 18:23:00:944 sip1(16328): SIPClientUserAgent Call using alternate outbound proxy of udp:69.59.142.213:5060.
DialPlan 18:23:00:944 sip1(16328): SIPClientUserAgent Call using alternate outbound proxy of udp:69.59.142.213:5060.
DialPlan 18:23:00:944 sip1(16328): SIPClientUserAgent Call using alternate outbound proxy of udp:69.59.142.213:5060.
DialPlan 18:23:00:944 sip1(16328): Switching to sip:inquba@187.133.147.183:49503 via udp:69.59.142.213:5060.
DialPlan 18:23:00:944 sip1(16328): Switching to sip:inquba@187.133.147.183:65176 via udp:69.59.142.213:5060.
DialPlan 18:23:00:944 sip1(16328): SDP on UAC call had public IP not mangled, RTP socket 64.154.41.150:19992.
DialPlan 18:23:00:944 sip1(16328): Switching to sip:inquba@187.133.147.183:65246 via udp:69.59.142.213:5060.
DialPlan 18:23:00:944 sip1(16328): SDP on UAC call had public IP not mangled, RTP socket 64.154.41.150:19992.
DialPlan 18:23:00:959 sip1(16328): SDP on UAC call had public IP not mangled, RTP socket 64.154.41.150:19992.
DialPlan 18:23:01:100 sip1(16328): Information response 100 Trying for sip:inquss@187.133.147.183:65246.
DialPlan 18:23:01:178 sip1(16328): Information response 100 Trying for sip:inquss@187.133.147.183:49503;rinstance=E09D708A;transport=tcp.
DialPlan 18:23:01:225 sip1(16328): Information response 180 Ringing for sip:inquss@187.133.147.183:65246.
DialPlan 18:23:01:225 sip1(16328): UAS call progressing with Ringing.
DialPlan 18:23:01:444 sip1(16328): Information response 180 Ringing for sip:inquss@187.133.147.183:49503;rinstance=E09D708A;transport=tcp.
DialPlan 18:23:01:444 sip1(16328): UAS call ignoring progress response with status of 180 as already in Proceeding.
DialPlan 18:23:01:678 sip1(16328): Information response 100 Trying for sip:inquss@187.133.147.183:65176.
DialPlan 18:23:01:959 sip1(16328): Information response 100 Trying for sip:inquss@187.133.147.183:65176.
DialPlan 18:23:02:006 sip1(16328): Information response 180 Ringing for sip:inquss@187.133.147.183:65176.
DialPlan 18:23:02:006 sip1(16328): UAS call ignoring progress response with status of 180 as already in Proceeding.

MikeTelis
Posts: 1582
Joined: Wed Jul 30, 2008 6:48 am

Re: **Error: Can't convert VSP into Fixnum

Post by MikeTelis » Fri May 13, 2011 6:44 pm

Uh-huh, apparently I can't find the right words to explain. You don't need to set @trunk, the dialplan will do that for you before passing control to incomingCall(). In the incomingCall you need to set Display name and/or User portion of the URI before you forward the call to bound SIP device(s). The code I posted here does not pass @trunk to ATA; you need to modify the code. If I was you, I'd try replacing fu=#{@cid} with fu=#{@trunk} and check how it works.
By The way the code for the test to gvny.inquss that is still showing the code 52 instead of +1 is this NOTE: the call was made for other mexican line and dial to my google voice abroad, the incoming call for all gv are ipcomms trunk that is config to forward calls to gvny.inquss@sipsorcery.com but in the console log is not any trace that the call is comming from gvny.inquss@sipsorcery.com and dont know why. since the trunk ipcomm is set correctly
No, the call is coming to inquss without any prefix. See for yourself:

:00:834 sip1(16328): ** Call from "3315920xxx" <sip:3315920766@64.154.41.150>;tag=as1f3d37f8 to inquss **

The dialplan uses +52 for all @trunk other than gvny and gvsm, that's why you get +52 instead of +1.

qubo
Posts: 127
Joined: Tue Apr 12, 2011 6:11 am

Re: **Error: Can't convert VSP into Fixnum

Post by qubo » Fri May 13, 2011 9:35 pm

thanks mike

I did your suggestion fu=#{@cid} with fu=#{@trunk} and check how it works. in console show that is comming from nextel fine but It Does Not show in the phone Display here the console

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DialPlan 21:29:59:457 sip1(1124): Using dialplan MEXICO for In call to sip:nextel.inquss@sipsorcery.com;rinstance=271355.
NewCall 21:29:59:472 sip1(1124): Executing script dial plan for call to nextel.inquss.
DialPlan 21:29:59:629 sip1(1124): ** Call from <sip:33159207xx@200.76.112.13:5060;user=phone>;tag=1fbe4e8-21724cc8-13c4-9145f-2f347db-9145f to nextel.inquss **
DialPlan 21:29:59:644 sip1(1124): Local time: 05/13/2011 16:29
DialPlan 21:29:59:660 sip1(1124): Caller's number: '5233159207xx'
DialPlan 21:29:59:722 sip1(1124): URI dialing: inquss@local[fu=nextel]
DialPlan 21:29:59:722 sip1(1124): Commencing Dial with: inquss@local[fu=nextel].
DialPlan 21:29:59:738 sip1(1124): Call leg is for local domain looking up bindings for inquss@sipsorcery.com for call leg inquss@local.
DialPlan 21:29:59:754 sip1(1124): 3 found for inquss@sipsorcery.com.
DialPlan 21:29:59:754 sip1(1124): ForkCall commencing call leg to sip:inquss@187.133.147.183:49181;rinstance=E09D708A;transport=tcp.
DialPlan 21:29:59:754 sip1(1124): ForkCall commencing call leg to sip:inquss@187.133.147.183:65246.
DialPlan 21:29:59:754 sip1(1124): SIPClientUserAgent Call using alternate outbound proxy of udp:69.59.142.213:5060.
DialPlan 21:29:59:754 sip1(1124): ForkCall commencing call leg to sip:inquss@187.133.147.183:65176.
DialPlan 21:29:59:754 sip1(1124): SIPClientUserAgent Call using alternate outbound proxy of udp:69.59.142.213:5060.
DialPlan 21:29:59:754 sip1(1124): Switching to sip:inquss@187.133.147.183:49181 via udp:69.59.142.213:5060.
DialPlan 21:29:59:754 sip1(1124): Switching to sip:inquss@187.133.147.183:65246 via udp:69.59.142.213:5060.
DialPlan 21:29:59:754 sip1(1124): SIPClientUserAgent Call using alternate outbound proxy of udp:69.59.142.213:5060.
DialPlan 21:29:59:754 sip1(1124): Switching to sip:inquss@187.133.147.183:65176 via udp:69.59.142.213:5060.
DialPlan 21:29:59:988 sip1(1124): Information response 100 Trying for sip:inquss@187.133.147.183:65246.
DialPlan 21:30:00:066 sip1(1124): Information response 100 Trying for sip:inquss@187.133.147.183:49181;rinstance=E09D708A;transport=tcp.
Uh-huh, apparently I can't find the right words to explain. You don't need to set @trunk, the dialplan will do that for you before passing control to incomingCall(). In the incomingCall you need to set Display name and/or User portion of the URI before you forward the call to bound SIP device(s). The code I posted here does not pass @trunk to ATA; you need to modify the code. If I was you, I'd try replacing fu=#{@cid} with fu=#{@trunk} and check how it works.

By The way the code for the test to gvny.inquss that is still showing the code 52 instead of +1 is this NOTE: the call was made for other mexican line and dial to my google voice abroad, the incoming call for all gv are ipcomms trunk that is config to forward calls to gvny.inquss@sipsorcery.com but in the console log is not any trace that the call is comming from gvny.inquss@sipsorcery.com and dont know why. since the trunk ipcomm is set correctly


No, the call is coming to inquss without any prefix. See for yourself:

:00:834 sip1(16328): ** Call from "3315920xxx" <sip:3315920766@64.154.41.150>;tag=as1f3d37f8 to inquss **

Code: Select all

yes I know that incoming call is arriving without the gvny even when the trunk is set to gvsm.inquss. maybe the server stuck with that update and is not correctly showing as should be. I will erase the trunk and then add again to see if the new create trunk would work as should be
NOTE UPDATE. it is so weird I erase the ipcomms trunks where Google Voice NUmber send the incoming calls. and even when I delete the trunks DIDs the call still arriving to inquss without prefix it is so weird it has been more than half hour since delete. I dont know if the the sipsorcery server is not updating changes here the console log

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DialPlan 21:54:36:355 sip1(9664): Using dialplan MEXICO for In call to sip:inquss@sipsorcery.com.
NewCall 21:54:36:371 sip1(9664): Executing script dial plan for call to inquss.
DialPlan 21:54:36:512 sip1(9664): ** Call from "unavailable" <sip:unavailable@64.154.41.150>;tag=as7330d868 to inquss **
DialPlan 21:54:36:512 sip1(9664): Local time: 05/13/2011 16:54
DialPlan 21:54:36:527 sip1(9664): Caller's number: 'unavailable'
DialPlan 21:54:36:558 sip1(9664): URI dialing: inquss@local[fu=]
DialPlan 21:54:36:558 sip1(9664): Commencing Dial with: inquss@local[fu=].
DialPlan 21:54:36:621 sip1(9664): Call leg is for local domain looking up bindings for inquss@sipsorcery.com for call leg inquss@local.
DialPlan 21:54:36:637 sip1(9664): 3 found for inquss@sipsorcery.com.
DialPlan 21:54:36:637 sip1(9664): ForkCall commencing call leg to sip:inquss@187.133.147.183:49181;rinstance=E09D708A;transport=tcp.
DialPlan 21:54:36:637 sip1(9664): ForkCall commencing call leg to sip:inquss@187.133.147.183:65246.
DialPlan 21:54:36:637 sip1(9664): ForkCall commencing call leg to sip:inquba@187.133.147.183:65176.
DialPlan 21:54:36:652 sip1(9664): SIPClientUserAgent Call using alternate outbound proxy of udp:69.59.142.213:5060.
DialPlan 21:54:36:652 sip1(9664): SIPClientUserAgent Call using alternate outbound proxy of udp:69.59.142.213:5060.
DialPlan 21:54:36:652 sip1(9664): Switching to sip:inquss@187.133.147.183:49181 via udp:69.59.142.213:5060.
DialPlan 21:54:36:652 sip1(9664): Switching to sip:inquss@187.133.147.183:65246 via udp:69.59.142.213:5060.
DialPlan 21:54:36:652 sip1(9664): SDP on UAC call had public IP not mangled, RTP socket 64.154.41.150:14836.
DialPlan 21:54:36:652 sip1(9664): SDP on UAC call had public IP not mangled, RTP socket 64.154.41.150:14836.
DialPlan 21:54:36:652 sip1(9664): SIPClientUserAgent Call using alternate outbound proxy of udp:69.59.142.213:5060.
DialPlan 21:54:36:652 sip1(9664): Switching to sip:inquss@187.133.147.183:65176 via udp:69.59.142.213:5060.
DialPlan 21:54:36:652 sip1(9664): SDP on UAC call had public IP not mangled, RTP socket 64.154.41.150:14836.
DialPlan 21:54:36:949 sip1(9664): Information response 100 Trying for 
The dialplan uses +52 for all @trunk other than gvny and gvsm, that's why you get +52 instead of +1.

MikeTelis

Posts: 1430
Joined: Wed Jul 30, 2008 12:48 am

MikeTelis
Posts: 1582
Joined: Wed Jul 30, 2008 6:48 am

Re: **Error: Can't convert VSP into Fixnum

Post by MikeTelis » Sat May 14, 2011 2:31 am

Are you trying to register to IPCOMMS? I've never done that and probably IPCOMMS doesn't support prefixes in Register Contact. Try this:

1. Go to http://sipconnect.ipcomms.net/ and login with your 10-digit DID phone number 305760xxxx and SIP password (also 10-digit). Note there's no spaces and other special chars in both login and password.
2. Click on DID in the menu on the left side.
3. Click on pencil icon in the "ACTION" column and change "Destination" to: SIP/GVNY.inquss@69.59.142.213 (SIP/GVNY.inquss@sipsorcery.com should also work).
4. Confirm changes.

qubo
Posts: 127
Joined: Tue Apr 12, 2011 6:11 am

Re: **Error: Can't convert VSP into Fixnum

Post by qubo » Sat May 14, 2011 5:42 am

thanks Mike yes the incoming DID for GV is ipcomm did I made what you told me and this time it works great. the weird thing is that I had already delete the ipcomm is not anymore in my sip provider and the call is arriving with the redirect gvny.inquss in the ipcomm site as your suggestion. and you are right ipcomm seems not work with prefix but your fix works. thanks
By the way did you see the previous question about make @trunk to display in the phone display? i modified what you said and the console detect the change in fu but the trunk is not displaying into the phone. I did your suggestion fu=#{@cid} with fu=#{@trunk} and check how it works. in console show that is comming from nextel fine but It Does Not show in the phone Display here the console
DialPlan 21:29:59:457 sip1(1124): Using dialplan MEXICO for In call to sip:nextel.inquss@sipsorcery.com;rinstance=271355.
NewCall 21:29:59:472 sip1(1124): Executing script dial plan for call to nextel.inquss.
DialPlan 21:29:59:629 sip1(1124): ** Call from <sip:33159207xx@200.76.112.13:5060;user=phone>;tag=1fbe4e8-21724cc8-13c4-9145f-2f347db-9145f to nextel.inquss **
DialPlan 21:29:59:644 sip1(1124): Local time: 05/13/2011 16:29
DialPlan 21:29:59:660 sip1(1124): Caller's number: '5233159207xx'
DialPlan 21:29:59:722 sip1(1124): URI dialing: inquss@local[fu=nextel]
DialPlan 21:29:59:722 sip1(1124): Commencing Dial with: inquss@local[fu=nextel].
DialPlan 21:29:59:738 sip1(1124): Call leg is for local domain looking up bindings for inquss@sipsorcery.com for call leg inquss@local.
DialPlan 21:29:59:754 sip1(1124): 3 found for inquss@sipsorcery.com.
DialPlan 21:29:59:754 sip1(1124): ForkCall commencing call leg to sip:inquss@187.133.147.183:49181;rinstance=E09D708A;transport=tcp.
DialPlan 21:29:59:754 sip1(1124): ForkCall commencing call leg to sip:inquss@187.133.147.183:65246.
DialPlan 21:29:59:754 sip1(1124): SIPClientUserAgent Call using alternate outbound proxy of udp:69.59.142.213:5060.
DialPlan 21:29:59:754 sip1(1124): ForkCall commencing call leg to sip:inquss@187.133.147.183:65176.
DialPlan 21:29:59:754 sip1(1124): SIPClientUserAgent Call using alternate outbound proxy of udp:69.59.142.213:5060.
DialPlan 21:29:59:754 sip1(1124): Switching to sip:inquss@187.133.147.183:49181 via udp:69.59.142.213:5060.
DialPlan 21:29:59:754 sip1(1124): Switching to sip:inquss@187.133.147.183:65246 via udp:69.59.142.213:5060.
DialPlan 21:29:59:754 sip1(1124): SIPClientUserAgent Call using alternate outbound proxy of udp:69.59.142.213:5060.
DialPlan 21:29:59:754 sip1(1124): Switching to sip:inquss@187.133.147.183:65176 via udp:69.59.142.213:5060.
DialPlan 21:29:59:988 sip1(1124): Information response 100 Trying for sip:inquss@187.133.147.183:65246.
DialPlan 21:30:00:066 sip1(1124): Information response 100 Trying for sip:inquss@187.133.147.183:49181;rinstance=E09D708A;transport=tcp.

MikeTelis
Posts: 1582
Joined: Wed Jul 30, 2008 6:48 am

Re: **Error: Can't convert VSP into Fixnum

Post by MikeTelis » Sat May 14, 2011 8:45 am

thanks Mike yes the incoming DID for GV is ipcomm did I made what you told me and this time it works great. the weird thing is that I had already delete the ipcomm is not anymore in my sip provider and the call is arriving with the redirect gvny.inquss in the ipcomm site as your suggestion. and you are right ipcomm seems not work with prefix but your fix works. thanks
You're welcome! SIP calls to your SIPaccount@sipsorcery.com will be processed, no matter you have the caller as your SIP provider or not. Generally, you need to add a VoIP provider as your SIP provider only if you want Sipsorcery server to register with this VoIP provider. It's convenient (but not absolutely necessary) to add SIP provider if you want to make outbound calls with it. As to incoming calls, if the provider can forward calls to your SIPaccount@sipsorcery.com, it's quite enough, you don't need to add it. I'm using IPCOMMS and IPKall DIDs in this manner, none of them are listed as my SIP providers.

qubo
Posts: 127
Joined: Tue Apr 12, 2011 6:11 am

Re: **Error: Can't convert VSP into Fixnum

Post by qubo » Sat May 14, 2011 8:56 am

thanks for all Mike all the dial plan you help me works Great thanks for all your patience and help I am very happy with your help. so this has finished with 10 ++++ plus thanks so much
By The Way. this is other question so far of this topic. but I want to take chance to ask you. I thin I have seen in other topics that you have experience with ATA devices so my question here is if you had have experience configuration linksys spa3000 as trunk or provider I was wondering and researching but until Now I could not find a solution. as you know spa3000 is a FXO pstn device. so I want to use as some trunk for outgoing calls only thanks for all

MikeTelis
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Joined: Wed Jul 30, 2008 6:48 am

Re: **Error: Can't convert VSP into Fixnum

Post by MikeTelis » Sat May 14, 2011 1:39 pm

qubo wrote:thanks for all Mike all the dial plan you help me works Great thanks for all your patience and help I am very happy with your help. so this has finished with 10 ++++ plus thanks so much
It's almost done :) Remember, the number formatting method is still sitting inside your dialplan while it belongs to 'mikesgem'? I have asked Aaron to update mikesgem on Sipsorcery server; once it's done, you can delete formatNum() method from your dialplan to make it work a little faster (and less load on the server).

UPDATE: Aaron has updated the 'mikesgem'; you can go ahead and remove formatNum() from your dialplan.
By The Way. this is other question so far of this topic. but I want to take chance to ask you. I thin I have seen in other topics that you have experience with ATA devices so my question here is if you had have experience configuration linksys spa3000 as trunk or provider I was wondering and researching but until Now I could not find a solution. as you know spa3000 is a FXO pstn device. so I want to use as some trunk for outgoing calls only thanks for all
I don't have any experience with SPA3000 (it's been obsoleted long time ago). I do own SPA3102 and I use its FXO port as SIP provider for my outbound calls within the city while I'm away from home. My IP address is dynamic; I had to employ DynDNS.com to solve this issue. Besides, I had to configure my router so that it would forward UDP packets to SIP and RTP ports to the SPA3102. Probably it can be done with SPA3000, too -- I just don't have it.

Try opening a new thread regarding this issue. We have quite a big community, someone might be able to help. I would also suggest posting your question onto the other relevant forums.

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