Search found 14 matches
- Fri Jan 26, 2018 9:08 pm
- Forum: Technical Support
- Topic: Could not find GALX key on your Google Voice pre-login page.
- Replies: 13
- Views: 9930
Re: Could not find GALX key on your Google Voice pre-login page.
Dear all, I am recently trying to use again GoogleVoice to call USA numbers. I only got these messages: DialPlan 20:53:36:045 sip1(7196): New call from udp:86.127.153.191:5060 successfully authenticated by digest. DialPlan 20:53:36:076 sip1(7196): Using dialplan default for Out call to sip:141690036...
- Fri Mar 18, 2011 8:51 am
- Forum: General VoIP Discussions
- Topic: How to insert Delay before dialing ?
- Replies: 1
- Views: 709
How to insert Delay before dialing ?
I want to insert a delay (20 seconds) before the phone will ring (incoming call). I have the following dial plan: #Ruby sys.Log("call from #{req.Header.From.FromURI.ToString()} to #{req.URI.User}.") if sys.In then # Do your incoming call processing customisations here. if sys.IsAvailable("#{req.URI....
- Tue Aug 17, 2010 5:32 am
- Forum: General VoIP Discussions
- Topic: Two ATAs behind home router ?
- Replies: 3
- Views: 810
Two ATAs behind home router ?
Hello! I have two SS account registered on two different ATAs located behind my home router, which has only one Public IP. One ATA is registered on 5060 port and the other on 5061. Also I set Stun as: stun.xten.net for both. I am able to initiate and to receive calls from each ATA only to outside (W...
- Mon Jun 07, 2010 5:29 pm
- Forum: Report a bug
- Topic: Silverlight error ?!
- Replies: 1
- Views: 505
Silverlight error ?!
Hi to all! I just upgraded my Silverlight to ver 4 as suggested when enter to sipsorcery.com But the are some errors: When I click to show my dial plan inside my account, the windows turn to black. It shows only : " Blog Forum Source Donate " and the rest of the window is black. All I can do is a re...
- Fri May 14, 2010 4:48 am
- Forum: Getting Started
- Topic: One stage calling using Linksys SPA 3102
- Replies: 0
- Views: 590
One stage calling using Linksys SPA 3102
Hi ! I have the folowing scenario: Location 1 - one SIP ATA Getek registered with Sipsorcery account (SS1) registered at VOIP prov account 1 (VSP1) (no PSTN line at this location) Location 2 - one Linksys SPA 3102 registered as follows - LINE1 (FXS port) reg.with SS2 account (VSP2) - PSTN (FXO -PSTN...
- Thu Jan 28, 2010 2:21 pm
- Forum: Getting Started
- Topic: Two stage calling ?
- Replies: 5
- Views: 1168
Hi !
You said that this script is for calling back to SipSorcery...but isn't suppose to exist a line for directing the call to SS? I do not see anything like "www.sipsorcery.com/webcallback..."
You said that this script is for calling back to SipSorcery...but isn't suppose to exist a line for directing the call to SS? I do not see anything like "www.sipsorcery.com/webcallback..."
- Wed Jan 27, 2010 2:34 pm
- Forum: Getting Started
- Topic: Two stage calling ?
- Replies: 5
- Views: 1168
Hi Aaron, and thanks for the answer. As I read on your Anouncement site here> http://sipsorcery.wordpress.com/2010/01/23/tropo-blind-transfers/ you manage some sort of transfer between Tropo and Sipsorcery. Being a newbe I did not understand all of your explanations there. So I present my Sipsorcery...
- Fri Jan 15, 2010 7:21 pm
- Forum: Getting Started
- Topic: Two stage calling ?
- Replies: 5
- Views: 1168
Two stage calling ?
Hello! I can use my Sipsorcery account to forward all my incomming calls to my mobile number if My ATA Device is not registered to Sipsorcery. Now I have a newbe question: Can sipsorcery provide like Two stage calling ? If someone call my DID registered in Sipsorcery , and my ATA is not registered, ...
- Wed Jun 24, 2009 8:12 pm
- Forum: Technical Support
- Topic: How to dial in SIP URI format
- Replies: 1
- Views: 473
How to dial in SIP URI format
For example how to call:
sip:8463@opalvoip.net via voxalot.com
using old format dial plan, because Ruby format seems to not work for me.
Thanks
sip:8463@opalvoip.net via voxalot.com
using old format dial plan, because Ruby format seems to not work for me.
Thanks
- Wed Jun 24, 2009 8:07 pm
- Forum: Feature Requests
- Topic: An automated test tools
- Replies: 0
- Views: 814
An automated test tools
Some numbers to call and to automaticaly get:
time or 8463 announce the current server time
addr or 2337 announce the clients (your) address
tone or 8063 play a three element tone
echo or 3246 enter an echo test
Thanks, Great Job !
HotM
time or 8463 announce the current server time
addr or 2337 announce the clients (your) address
tone or 8063 play a three element tone
echo or 3246 enter an echo test
Thanks, Great Job !
HotM