Click Here To Visit SIP Broker  

Go Back   Voxalot / SIP Broker Support Forums > Voxalot Forums > Voxalot Support

Voxalot Support Support for the Voxalot service.

 
 
Reply
Thread Tools Display Modes
Unread 12-29-2007, 10:15 PM   #21
Kenthurst35
Junior Member
 
Join Date: Dec 2007
Posts: 26
Thanks: 0
Thanked 1 Time in 1 Post
Kenthurst35 is on a distinguished road
Default

This config is currently set up for routing via the PSTN gateway.

I did have those CFwd parameters on Line 1 set to YES originally, but they didn't seem to have any effect since I was using the gateway method to divert to VM.

Happy to reset them to YES if you think it will make a difference.

Cheers
Kenthurst35 is offline   Reply With Quote
Unread 12-29-2007, 10:16 PM   #22
Kenthurst35
Junior Member
 
Join Date: Dec 2007
Posts: 26
Thanks: 0
Thanked 1 Time in 1 Post
Kenthurst35 is on a distinguished road
Default

BTW, I don't give out my VoIP DID, so inbound calls are always via PSTN.

Cheers
Kenthurst35 is offline   Reply With Quote
Unread 12-29-2007, 10:26 PM   #23
dc2007
Member
 
Join Date: Mar 2007
Posts: 81
Thanks: 12
Thanked 1 Time in 1 Post
dc2007 is on a distinguished road
Default

can you make your vm ringing duration lower than your spa3102 PSTN Answer Delay?
dc2007 is offline   Reply With Quote
Unread 12-29-2007, 10:37 PM   #24
Kenthurst35
Junior Member
 
Join Date: Dec 2007
Posts: 26
Thanks: 0
Thanked 1 Time in 1 Post
Kenthurst35 is on a distinguished road
Default

The current settings are:

In the Voxalot account, voicemail ringing duration is set to 5 secs. In the SPA3102 the PSTN Answer Delay is set to 10 secs.

With these settings, my understanding of the call flow is this:

PSTN call --> Line 1 --> ring for duration set in PSTN Answer Delay (10) --> gateway takes call & dials voxalot account --> ring for Voxalot voicemail ringing duration (5) --> VM

Cheers
Kenthurst35 is offline   Reply With Quote
Unread 01-02-2008, 09:14 AM   #25
Kenthurst35
Junior Member
 
Join Date: Dec 2007
Posts: 26
Thanks: 0
Thanked 1 Time in 1 Post
Kenthurst35 is on a distinguished road
Default

Anyone out there, with some more ideas on what could be the problem here?

I really am at my wits end with this.
Kenthurst35 is offline   Reply With Quote
Unread 01-04-2008, 01:16 PM   #26
ozimarco
Senior Member
 
Join Date: May 2006
Location: Lower Chittering, about an hour from Perth, Western Australia
Posts: 229
Thanks: 61
Thanked 42 Times in 28 Posts
ozimarco is on a distinguished road
Send a message via Skype™ to ozimarco
Default

Quote:
Originally Posted by Kenthurst35 View Post
Anyone out there, with some more ideas on what could be the problem here?

I really am at my wits end with this.
I have one more thing for you to try. It solved some problems for me but I'm not sure it is related to your problem. Enable STUN (I use stun.voxalot.com:3478), enable NAT mapping and NAT keep alive. Disable any port forwarding you have in place between the ATA and the router.

Might pay to re-boot everything as well.
__________________
ISP: Internode Easy Reach 60: 60GB for AUD60
VSPs: Pennytel and others via SIPSorcery
Hardware:
Billion 7800VDPX + Siemens Gigaset A580IP and Yealink T22P.

Last edited by ozimarco; 01-04-2008 at 01:19 PM.
ozimarco is offline   Reply With Quote
Unread 01-05-2008, 03:09 AM   #27
Kenthurst35
Junior Member
 
Join Date: Dec 2007
Posts: 26
Thanks: 0
Thanked 1 Time in 1 Post
Kenthurst35 is on a distinguished road
Default

Thanks Ozimarco

I'll try using the STUN settings as you suggest and see what happens.

Thing is, if I can make SIP calls to other addresses successfully, wouldn't that indicate that I don't have a STUN problem?

Cheers
Kenthurst35 is offline   Reply With Quote
Unread 01-05-2008, 03:19 AM   #28
Kenthurst35
Junior Member
 
Join Date: Dec 2007
Posts: 26
Thanks: 0
Thanked 1 Time in 1 Post
Kenthurst35 is on a distinguished road
Default

OK, I tried changing the STUN settings, but no joy.

Here is the syslog trace:

syslog server(port:514) started on Sat Jan 05 14:11:46 2008
TP Parser error: 34
FXO:Start CNDD
Calling:201259@127.0.0.1:5060
[1:0]AUD ALLOC CALL (port=16384)
[1:0]RTP Rx Up
[0:0]AUD ALLOC CALL (port=16386)
[0:0]RTP Rx Up
CC:Ringback
AUD:Play PSTN Tone 9
[1:0]RTP Rx Dn
FXO:CNDD name=, number=
FXO:Stop CNDD
FXO:CNDD Name= Phone=
AUD:Stop PSTN Tone
[5062]STUN trying 0
[16388]STUN trying 0
[16389]STUN trying 0
[16390]STUN trying 0
[16391]STUN trying 0
AUD:Stop PSTN Tone
[0]FM Alert Stop RxTx (c=0024eaa4;a=0)
[1:0]AUD Rel Call
[0]FM Alert Stop RxTx (c=0024900c;a=0)
[0:0]AUD Rel Call
CC:Ended
DLG Terminated 2cf2f8
[1:0]CC:STUN OK:c0a80196->7cbcf270, 5062->1567 16388->16388
AUD:Stop PSTN Tone
Calling:201259@au.voxalot.com:0
[1:0]AUD ALLOC CALL (port=16388)
[1:0]RTP Rx Up
DLG Terminated 2cf264
Sess Terminated
Sess Terminated
TP Parser error: 34
AUD:Stop PSTN Tone
FXO:On Hook
AUD:Stop PSTN Tone
FXO:Stop CNDD
[0]FM Alert Stop RxTx (c=0024eaa4;a=0)
[1:0]AUD Rel Call
TP Parser error: 34

I've tried this with another Voxalot account I set up, but same result. Other SIP addresses seem to call OK.

Cheers
Kenthurst35 is offline   Reply With Quote
Unread 01-05-2008, 04:46 AM   #29
Kenthurst35
Junior Member
 
Join Date: Dec 2007
Posts: 26
Thanks: 0
Thanked 1 Time in 1 Post
Kenthurst35 is on a distinguished road
Default Fwd SPA3102 PST calls to Vox Voicemail {solved}

I finally got this to work by changing the following:
  1. Enabling STUN server (thanks for the tip Ozimarco)
  2. Setting Use DNS SRV on the PSTN tab to YES (this solves the GetServerAddrErr message appearing in syslog)
  3. On the SIP tab, setting Substitute VIA Addr to YES and Send Resp To SRC Port to YES
  4. In my Voxalot account, setting a Call Connection rule that anyone calling my account gets diverted to VM. I also have Symmetric NAT Handling set to No in my Voxalot account

The call routing behaviour I now have is that on Line 1 busy or no answer, my calls go th Voxalot VM. Also, because I have call waiting active on Line 1, the user has the option of picking up the call before it goes to Voxalot.

Note: all CFwd fields on User tab are blank.

Thanks to everyone who contributed to this thread.

Cheers
Kenthurst35 is offline   Reply With Quote
Unread 03-20-2008, 12:20 PM   #30
gotalkvoip
Junior Member
 
Join Date: Mar 2008
Posts: 28
Thanks: 4
Thanked 0 Times in 0 Posts
gotalkvoip is on a distinguished road
Default

Just couple questions to Kenthurst35, just wondering which VSP you using? and I sought of have the same problem but mine is any private callers calling wont allow my VSP to forward the call to my mobile,. wonder if private calls foward ok on your setup?

cheerz

ps: anyway glad you've solved your problem :-)

Last edited by gotalkvoip; 03-20-2008 at 12:32 PM.
gotalkvoip is offline   Reply With Quote
Reply

Thread Tools
Display Modes

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off
Forum Jump

Similar Threads
Thread Thread Starter Forum Replies Last Post
Free DID Numbers, Free Voip Calls, & more Voip Info amroe Voxalot General 54 01-13-2014 09:11 AM
incoming sip calls going directly to voicemail vishal3110 Voxalot Support 8 12-13-2008 12:17 AM
X-Lite and sending calls to voicemail jonfry Voxalot Support 1 04-11-2007 01:44 AM
Codec order for SIP - PSTN calls occamsrazor Voxalot Support 0 03-08-2007 10:05 AM
Forward calls from PSTN to mobile via Pennytel - HELP BJReplay Voxalot Support 0 12-22-2006 12:04 AM


All times are GMT. The time now is 03:46 AM.


Powered by vBulletin® Version 3.7.2
Copyright ©2000 - 2019, Jelsoft Enterprises Ltd.