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Unread 10-17-2006, 05:29 PM   #1
code-
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Default What am I doing wrong here?

Been doing some testing, and I am having a problem dialing intl. calls.
My dial plans:
Code:
_xxxxxxxx	0047${EXTEN}	voipbuster.com (Norway, works 100%)
_1.		00${EXTEN}	voipbuster.com (USA/Canada, not working)
_00.		${EXTEN}	voipbuster.com (All others, not working)
If I dial 12345678 (first rule) it goes through fine. However, dialing 0047-12345678 (last rule) does NOT work...

If I dial 001-xxx-xxx-xxxx or 1-xxx-xxx-xxxx for example the call does not go through.
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Unread 10-17-2006, 06:09 PM   #2
code-
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Here are some logs from the ATA, hope this can help diagnose the problem:

Working xxxxxxxx call:
Code:
000040835 EPTAPI: _ept_event_cb: line 1 off-hook
000040835 SIP receive event: ept_id = 0, event = 1
000040835 task_sipmain: ept_id = 0, event = 1
000040835 task_sipmain: line1: SipEventOffHook
000040835 EPTAPI: Line 1 ept_phone_hook_off  hook_state=OFFHOOK
000040835 SIP: [Line 1] hook state changed OnHook ---> OffHook
000040835 Line 1 -- play dial tone.
000040835 SIP: [Line 1] call state changed StateNull ---> StatePending
000041673 New SIP session created(sd=1).
000041673 SIP: [Line 1] call state changed StatePending ---> StateOriginated
000041673 SIP receive event: ept_id = 0, event = 52,reason = 1
000041673 task_sipmain: ept_id = 0, event = 52
000041673 SIP: sipSendRequst session=1 method=1
000041673 Line 1 --- Send SIP Message to 64.34.163.35:5060 (sd=1, size=849)
000041689 Line 1 --- Receive SIP Message from 64.34.163.35:5060 (size=528)
000041689 SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.191.54.34:5060;branch=z9hG4bK53cf0f0c87729566
From: 551082 <sip:551082@voxalot.com;user=phone>;tag=0ee818980d97e2f9
To: <sip:75007503@voxalot.com;user=phone>
Call-ID: d9a9c5d5cb734144@81.191.54.34
CSeq:000041689 SIP: [Line 1] call state changed StateOriginated ---> StateDelivered
000041705 Line 1 --- Receive SIP Message from 64.34.163.35:5060 (size=703)
000041705 SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 81.191.54.34:5060;branch=z9hG4bK53cf0f0c87729566
From: 551082 <sip:551082@voxalot.com;user=phone>;tag=0ee818980d97e2f9
To: <sip:75007503@voxalot.com;user=phone>;tag=1740dee5c04b38e1fba000041705 SIP: sipSendRequst session=1 method=2
000041705 SIP: sipSendRequst session=1 method=1
000041707 Line 1 --- Send SIP Message to 64.34.163.35:5060 (sd=1, size=406)
000041711 Line 1 --- Send SIP Message to 64.34.163.35:5060 (sd=1, size=1055)
000041727 Line 1 --- Receive SIP Message from 64.34.163.35:5060 (size=528)
000041727 SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.191.54.34:5060;branch=z9hG4bK59af51235d7ede9c
From: 551082 <sip:551082@voxalot.com;user=phone>;tag=0ee818980d97e2f9
To: <sip:75007503@voxalot.com;user=phone>
Call-ID: d9a9c5d5cb734144@81.191.54.34
CSeq:000041727 SIP: [Line 1] call state changed StateDelivered ---> StateDelivered
000041745 Line 1 --- Receive SIP Message from 64.34.163.35:5060 (size=572)
000041745 SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 81.191.54.34:5060;branch=z9hG4bK59af51235d7ede9c
From: 551082 <sip:551082@voxalot.com;user=phone>;tag=0ee818980d97e2f9
To: <sip:75007503@voxalot.com;user=phone>
Call-ID: d9a9000041745 SIP: [Line 1] call state changed StateDelivered ---> StateDelivered
000041759 Line 1 --- Receive SIP Message from 64.34.163.35:5060 (size=769)
000041759 SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 81.191.54.34:5060;branch=z9hG4bK59af51235d7ede9c
From: 551082 <sip:551082@voxalot.com;user=phone>;tag=0ee818980d97e2f9
To: <sip:75007503@voxalot.com;user=phone>;tag=as520e7965
Call-ID: d9a9c5d5cb7000041759 SDP decode attribute token=PCMU, ret=0
000041759 SDP decode attribute token=PCMA, ret=8
000041759 SDP decode attribute token=G729, ret=18
000041759 SDP decode attribute token=telephone-event, ret=97
000041759 SIP: [Line 1] call state changed StateDelivered ---> StateDelivered
000041759 psipSession->state != SipStateConnected!
000041759 Check Msg for psipSession->sdp_changed=1
000041759 sipSetRemoteSDP() ptime=20
000041759 SIP receive event: ept_id = 0, event = 71,reason = 1
000041759 task_sipmain: ept_id = 0, event = 71
000041759 SDP Change! Call sipConnect()!!
000041759 SIP: Line 1 sipConnect() sd=1 remote_adderss=64.34.163.35:16398 connection_mode=3
000041759 SIP: Line 1 sipConnect() codec=0 oob_dtmf=1 payload=97
000041760 EPTAPI: LINE 1 ept_set_phone_volume  txgain = -11  rxgain = -3  ok
000041760 EPTAPI: Line 1 ept_ec_set  enable=1
000041760 EPTAPI: Line 1 ept_ec_set  enable=1 ec_level=2
000041760 EPTAPI: Line 1 nLen=16  bNlp=1   nGainIn=2  nGainOut=2
000041760 EPTAPI: Line 1 ept_vad_set vad_value=2 vad_level=1
000041760 EPTAPI: Line 1 ept_vad_set vad_value=4
000041760 EPTAPI: ept_set_jitter_buffer = enable ok
000041760 EPTAPI: codec_channel_id 0 ept_set_frame_length  len = 20  ok
000041760 0 remote_payload =97,1, local_payload=97,1
000041760 EPTAPI: Line 1 ept_rtp_configuration  RtpParam.nEvents=2 RtpParam.nEventPT=97
000041760 EPTAPI: Line 1 ept_rtp_modify  remote_ip=64.34.163.35:16398
000041760 VINETIC_RTP: vinetic_rtp_configure_port  Create RTP socket 47
000041760 VINETIC_RTP: vinetic_rtp_configure_port FD=47 remote_ip=64.34.163.35:16398
000042267 Line 1 --- Receive SIP Message from 64.34.163.35:5060 (size=817)
000042267 SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.191.54.34:5060;branch=z9hG4bK59af51235d7ede9c
Record-Route: <sip:64.34.163.35;lr=on;ftag=0ee818980d97e2f9>
From: 551082 <sip:551082@voxalot.com;user=phone>;tag=0ee818980d97e2f9
To: <sip:75007503@voxalot.com;000042267 SDP decode attribute token=PCMU, ret=0
000042267 SDP decode attribute token=PCMA, ret=8
000042267 SDP decode attribute token=G729, ret=18
000042267 SDP decode attribute token=telephone-event, ret=97
000042267 psipSession->state != SipStateConnected!
000042267 Check Msg for psipSession->sdp_changed=1
000042267 sipSetRemoteSDP() ptime=20
000042267 SIP: [Line 1] call state changed StateDelivered ---> StateEstablished
000042267 SIP: [Line 1] start Outgoing call
000042267 SIP receive event: ept_id = 0, event = 10
000042267 ###1st 200 OK session timer:0###
000042267 Line 1: Connected. Codec=0
000042267 SIP: sipSendRequst session=1 method=2
000042268 SIP receive event: ept_id = 0, event = 71,reason = 1
000042268 Line 1: Re-Invite. Codec=0
000042268 <SipCallTransfer(All)>:Call Connected, set sipInfo[1].ept_info.dtmf_buf[0] = 0.
000042268 task_sipmain: ept_id = 0, event = 10
000042268 task_sipmain: line 1 --- stop ringback.
000042268 task_sipmain: ept_id = 0, event = 71
000042268 SDP Change! Call sipConnect()!!
000042268 SIP: Line 1 sipConnect() sd=1 remote_adderss=64.34.163.35:16398 connection_mode=3
000042268 SIP: Line 1 sipConnect() codec=0 oob_dtmf=1 payload=97
000042269 Line 1 --- Send SIP Message to 64.34.163.35:5060 (sd=1, size=447)
000043981 EPTAPI: _ept_event_cb: line 1 on-hook
000043981 SIP receive event: ept_id = 0, event = 2
000043981 task_sipmain: ept_id = 0, event = 2
000043981 task_sipmain: line1: SipEventOnHook
000043981 SIP: [Line 1] stop Outgoing call
000043981 SIP: [Line 1] hook state changed OffHook ---> OnHook
000043981 SIP: [Line 1] call state changed StateEstablished ---> StateNull
000043981 bye:1
000043981 SIP: sipSendRequst session=1 method=3
000043981 Start close wait timer for session 1.
000043981 SIP receive event: ept_id = 0, event = 73,reason = 1
000043981 task_sipmain: ept_id = 0, event = 73
000043981 SIP: Line 1 sipDisconnect() sd=1 all=1
000043981 EPTAPI: Line 1 ept_rtp_off  sd=1 delete ALL
000043981 EPTAPI: Line 1  ept_rtp_off MATCH  codec_channel_id=0
000043981 EPTAPI: Codec_channel_id  0 ept_rtp_free_port_specific
000043983 Line 1 --- Send SIP Message to 64.34.163.35:5060 (sd=1, size=510)
000043997 Line 1 --- Receive SIP Message from 64.34.163.35:5060 (size=538)
000043997 SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.191.54.34:5060;branch=z9hG4bK035f58aedc9c1191
Record-Route: <sip:64.34.163.35;lr=on;ftag=0ee818980d97e2f9>
From: 551082 <sip:551082@voxalot.com;user=phone>;tag=0ee818980d97e2f9
To: <sip:75007503@voxalot.com;000043997 === Close session 1
000043997 Session 1 closed.
000043997 Line 1: Connection closed.
Non-working call to the same number but with 0047 in front:
Code:
000064409 EPTAPI: _ept_event_cb: line 1 off-hook
000064409 SIP receive event: ept_id = 0, event = 1
000064409 task_sipmain: ept_id = 0, event = 1
000064409 task_sipmain: line1: SipEventOffHook
000064409 EPTAPI: Line 1 ept_phone_hook_off  hook_state=OFFHOOK
000064409 SIP: [Line 1] hook state changed OnHook ---> OffHook
000064409 Line 1 -- play dial tone.
000064409 SIP: [Line 1] call state changed StateNull ---> StatePending
000065319 New SIP session created(sd=1).
000065319 SIP: [Line 1] call state changed StatePending ---> StateOriginated
000065319 SIP receive event: ept_id = 0, event = 52,reason = 1
000065319 task_sipmain: ept_id = 0, event = 52
000065319 SIP: sipSendRequst session=1 method=1
000065319 Line 1 --- Send SIP Message to 64.34.163.35:5060 (sd=1, size=859)
000065335 Line 1 --- Receive SIP Message from 64.34.163.35:5060 (size=540)
000065335 SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.191.54.34:5060;branch=z9hG4bKb79d9e83fde3e6c3
From: 551082 <sip:551082@voxalot.com;user=phone>;tag=1969fc91e1093557
To: <sip:004775007503@voxalot.com;user=phone>
Call-ID: daf1f01573bc1b5c@81.191.54.34
C000065335 SIP: [Line 1] call state changed StateOriginated ---> StateDelivered
000065371 Line 1 --- Receive SIP Message from 64.34.163.35:5060 (size=715)
000065371 SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 81.191.54.34:5060;branch=z9hG4bKb79d9e83fde3e6c3
From: 551082 <sip:551082@voxalot.com;user=phone>;tag=1969fc91e1093557
To: <sip:004775007503@voxalot.com;user=phone>;tag=1740dee5c04b38e000065371 SIP: sipSendRequst session=1 method=2
000065371 SIP: sipSendRequst session=1 method=1
000065373 Line 1 --- Send SIP Message to 64.34.163.35:5060 (sd=1, size=414)
000065377 Line 1 --- Send SIP Message to 64.34.163.35:5060 (sd=1, size=1069)
000065393 Line 1 --- Receive SIP Message from 64.34.163.35:5060 (size=540)
000065393 SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.191.54.34:5060;branch=z9hG4bKe06190cac4e5d36d
From: 551082 <sip:551082@voxalot.com;user=phone>;tag=1969fc91e1093557
To: <sip:004775007503@voxalot.com;user=phone>
Call-ID: daf1f01573bc1b5c@81.191.54.34
C000065393 SIP: [Line 1] call state changed StateDelivered ---> StateDelivered
000065425 Line 1 --- Receive SIP Message from 64.34.163.35:5060 (size=584)
000065425 SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 81.191.54.34:5060;branch=z9hG4bKe06190cac4e5d36d
From: 551082 <sip:551082@voxalot.com;user=phone>;tag=1969fc91e1093557
To: <sip:004775007503@voxalot.com;user=phone>
Call-ID: 000065425 SIP: [Line 1] call state changed StateDelivered ---> StateDelivered
000065459 Line 1 --- Receive SIP Message from 64.34.163.35:5060 (size=777)
000065459 SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 81.191.54.34:5060;branch=z9hG4bKe06190cac4e5d36d
From: 551082 <sip:551082@voxalot.com;user=phone>;tag=1969fc91e1093557
To: <sip:004775007503@voxalot.com;user=phone>;tag=as02ee9daf
Call-ID: daf1f01000065459 SDP decode attribute token=PCMU, ret=0
000065459 SDP decode attribute token=PCMA, ret=8
000065459 SDP decode attribute token=G729, ret=18
000065459 SDP decode attribute token=telephone-event, ret=97
000065459 SIP: [Line 1] call state changed StateDelivered ---> StateDelivered
000065459 psipSession->state != SipStateConnected!
000065459 Check Msg for psipSession->sdp_changed=1
000065459 sipSetRemoteSDP() ptime=20
000065459 SIP receive event: ept_id = 0, event = 71,reason = 1
000065459 task_sipmain: ept_id = 0, event = 71
000065459 SDP Change! Call sipConnect()!!
000065459 SIP: Line 1 sipConnect() sd=1 remote_adderss=64.34.163.35:19390 connection_mode=3
000065460 SIP: Line 1 sipConnect() codec=0 oob_dtmf=1 payload=97
000065460 EPTAPI: LINE 1 ept_set_phone_volume  txgain = -11  rxgain = -3  ok
000065460 EPTAPI: Line 1 ept_ec_set  enable=1
000065460 EPTAPI: Line 1 ept_ec_set  enable=1 ec_level=2
000065460 EPTAPI: Line 1 nLen=16  bNlp=1   nGainIn=2  nGainOut=2
000065460 EPTAPI: Line 1 ept_vad_set vad_value=2 vad_level=1
000065460 EPTAPI: Line 1 ept_vad_set vad_value=4
000065460 EPTAPI: ept_set_jitter_buffer = enable ok
000065460 EPTAPI: codec_channel_id 0 ept_set_frame_length  len = 20  ok
000065460 0 remote_payload =97,1, local_payload=97,1
000065460 EPTAPI: Line 1 ept_rtp_configuration  RtpParam.nEvents=2 RtpParam.nEventPT=97
000065460 EPTAPI: Line 1 ept_rtp_modify  remote_ip=64.34.163.35:19390
000065460 VINETIC_RTP: vinetic_rtp_configure_port  Create RTP socket 26
000065460 VINETIC_RTP: vinetic_rtp_configure_port FD=26 remote_ip=64.34.163.35:19390

--- After about 1 min of silence, I hang up ---

000073644 EPTAPI: _ept_event_cb: line 1 on-hook
000073644 SIP receive event: ept_id = 0, event = 2
000073644 task_sipmain: ept_id = 0, event = 2
000073644 task_sipmain: line1: SipEventOnHook
000073644 SIP: [Line 1] hook state changed OffHook ---> OnHook
000073644 SIP: [Line 1] call state changed StateDelivered ---> StateNull
000073644 SIP: sipSendRequst session=1 method=5
000073644 Start close wait timer for session 1.
000073644 SIP receive event: ept_id = 0, event = 73,reason = 1
000073644 task_sipmain: ept_id = 0, event = 73
000073644 SIP: Line 1 sipDisconnect() sd=1 all=1
000073644 EPTAPI: Line 1 ept_rtp_off  sd=1 delete ALL
000073644 EPTAPI: Line 1  ept_rtp_off MATCH  codec_channel_id=0
000073644 EPTAPI: Codec_channel_id  0 ept_rtp_free_port_specific
000073644 SET VOICE OFF
000073645 Line 1 --- Send SIP Message to 64.34.163.35:5060 (sd=1, size=378)
000073695 SIP receive event: ept_id = 0, event = 76,reason = 1
000073695 task_sipmain: ept_id = 0, event = 76
000073695 Line 1 --- Retransmit SIP Message to 64.34.163.35:5060 (sd=1, size=378)
000073795 SIP receive event: ept_id = 0, event = 76,reason = 1
000073795 task_sipmain: ept_id = 0, event = 76
000073795 Line 1 --- Retransmit SIP Message to 64.34.163.35:5060 (sd=1, size=378)
000073995 SIP receive event: ept_id = 0, event = 76,reason = 1
000073995 task_sipmain: ept_id = 0, event = 76
000073995 Line 1 --- Retransmit SIP Message to 64.34.163.35:5060 (sd=1, size=378)
000074395 SIP receive event: ept_id = 0, event = 76,reason = 1
000074395 task_sipmain: ept_id = 0, event = 76
000074395 Line 1 --- Retransmit SIP Message to 64.34.163.35:5060 (sd=1, size=378)
000074795 SIP receive event: ept_id = 0, event = 76,reason = 1
000074795 task_sipmain: ept_id = 0, event = 76
000074795 Line 1 --- Retransmit SIP Message to 64.34.163.35:5060 (sd=1, size=378)
000075195 SIP receive event: ept_id = 0, event = 76,reason = 1
000075195 task_sipmain: ept_id = 0, event = 76
000075195 Line 1 --- Retransmit SIP Message to 64.34.163.35:5060 (sd=1, size=378)
000075595 SIP receive event: ept_id = 0, event = 76,reason = 1
000075595 task_sipmain: ept_id = 0, event = 76
000075595 Line 1 --- Retransmit SIP Message to 64.34.163.35:5060 (sd=1, size=378)
000075995 SIP receive event: ept_id = 0, event = 76,reason = 1
000075995 task_sipmain: ept_id = 0, event = 76
000075995 Line 1 --- Retransmit SIP Message to 64.34.163.35:5060 (sd=1, size=378)
000076395 SIP receive event: ept_id = 0, event = 76,reason = 1
000076395 task_sipmain: ept_id = 0, event = 76
000076395 Line 1 --- Retransmit SIP Message to 64.34.163.35:5060 (sd=1, size=378)
000076795 SIP receive event: ept_id = 0, event = 76,reason = 1
000076795 task_sipmain: ept_id = 0, event = 76
000076795 Line 1 --- Retransmit SIP Message to 64.34.163.35:5060 (sd=1, size=378)
000076844 === Close session 1
000076844 Session 1 closed.
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Unread 10-17-2006, 09:49 PM   #3
Jorge
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For what I understand, without looking at your logs, it could be like this:

Code:
_xxxxxxxx	0047${EXTEN}	voipbuster.com
_001X.		${EXTEN:2}	voipbuster.com
_00X.		${EXTEN}	voipbuster.com
This way, in rule 2 the leading 00 is stripped, I assume your provider expects a leading 1 because otherwise rule 3 would cover this cae.

Without the X, maybe your rules expect more ones or zeros!
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Unread 10-17-2006, 10:12 PM   #4
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As Jorge says you need a X in there. For example _1. means one or more 1's while _1X. means a numbers beginning with '1' followed by one or more digits.

Hope that makes sense?
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Unread 10-17-2006, 10:54 PM   #5
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Hey, I had this exact same problem and at first I thought it was my provider. You can see my thread at this link:

http://forum.voxalot.com/showthread.php?t=521

I found that you had to put any thing starting with an _X as the very last priority. You also noticed that the call tester (the thing where you see which provider calls are being routed to) did not pick up on this. It was saying my calls would transfered the way I wanted, however, they were not. Put the _xxxxxxx as your last priority, and that should fix it.

After reading what jorge said, he is right. I did not realize what exactly was happening, but I think both ways will work though.

Last edited by klydal; 10-17-2006 at 10:58 PM.
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Unread 10-17-2006, 11:46 PM   #6
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Quote:
Originally Posted by martin View Post
As Jorge says you need a X in there. For example _1. means one or more 1's while _1X. means a numbers beginning with '1' followed by one or more digits.
Martin,

That's the way it works in a Linksys/Sipura dial plan, but not in Asterisk. The example given on the VoXaLot Dial Plan entry screen is:

. wildcard, matches one or more characters

e.g. _9. matches 94811111 etc

I have two Dial Plan entrys set up this way and they seem to work properly:

_7. ${EXTEN:1} Free World Dialup
_0011. ${EXTEN:1} Gizmo Project

In each case, I want the leading digit stripped and the remaining digits passed to the provider.

Ron
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Unread 10-18-2006, 12:07 AM   #7
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Your right Ron I stand corrected. Sorry for misleading you code-
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Unread 10-18-2006, 03:16 AM   #8
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Quote:
Originally Posted by code- View Post
If I dial 12345678 (first rule) it goes through fine. However, dialing 0047-12345678 (last rule) does NOT work...

If I dial 001-xxx-xxx-xxxx or 1-xxx-xxx-xxxx for example the call does not go through.
What are you using for an ATA or softphone connected to VoXaLot? I suspect you have a problem in its dial plan and the number you are entering on the phone is not reaching VoXaLot in the format you think it is.

Ron
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Unread 10-18-2006, 09:21 AM   #9
code-
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I am using a D-Link DVG-G1402S VoIP Wireless Router

You can see that in the logs, this is working:
To: <sip:75007503@voxalot.com> (My first rule adds 0047 and sends til voipbuster)

And this is not:
To: <sip:004775007503@voxalot.com> (My last rule should send this directly til voipbuster, and says it will)

According to the VoXaLot web interface both of these nubmers are routed to exactly the same sip address.

Is there any way to get debugging output from the VoXaLot server?
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Unread 10-18-2006, 07:45 PM   #10
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OK, something is very wrong with the way the dial plans are prosessed.
New order:
Code:
0	_0x.		${EXTEN}	voipbuster.com	
1	_xxxxxxxx	0047${EXTEN}	voipbuster.com		
2	_1xxxxxxxxxx	00${EXTEN}	voipbuster.com
Calling 0047-12345678 (0), 12345678 (1) and 001-123-456-7890 (1) is now working, but 1-xxx-xxx-xxxx (2, last rule) is not.
Changing the order should not have made any difference, but it did. Why?
And how does the "test" in the dial plan list page differ from the actual prosessing done by Asterisk?
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