 Menu Home News Products Services H.239 Buy Now Support FAQ Projects Download Press about us Contacts About us PSGw testimonial: "...I am delighted to announce that the interface between Skype and SIP works like a charm, and at the remarkably good voice quality often reported by Skype users. We run PSGw with single Skype account to receive calls and route them to our telephone operator, to be forwarded on our Asterisk PBX to the intended recipient (or conference box). Source: http://zog.typepad.com Read all testimonials |
Frequently asked questions1. PSGw configuration looks complex. Why?2. Where can I get more info about the PSGw "Routing table"? I need more information regarding PSGw rules. 3. SIP Basics 4. Got "404 Error" at the SIP endpoint when placing a call from a SIP device to PSGw 5. OK, I can call "echo123". How can I connect to another Skype user? What should I specify as "Outgoing username"? 6. Hmm, PSGw doesn't connect to the SIP gateway when I call PSGw from Skype. I see: "SIP/2.0 404 Not Found" in the SIP device log. 7. Does PSGw relay or bypass DTMF tones? 8. Can I run PSGw in VMWare? 9. Can I run PSGw remotelly using Remote Desktop? 10. Are there any examples of SIP configuration for well-known SIP providers? 11. I run Skype and PSGw. Skype is logged in with the "myusername" username. Can I call my Skype "myusername" with a SIP or H.323 phone? 12. I run Skype and PSGw. Skype is logged in with the "myusername" username. Can I use my local Skype to call SIP or H.323 phones? 13. I use PSGw with Asterisk. Sometimes Asterisk shows "481 Call Leg/Transaction Does Not Exist" message. Why? 14. I want to use PSGw with Asterisk. How should I configure PSGw? 15. How must I configure Skype to use it with PSGw? Are there any specific requirements? 1. PSGw configuration looks complex. Why? Originally PSGw was designed to be user-friendly and easy-configurable. However knowledge of SIP essentials is necessary. If the "Registrar" and "Proxy" terms sound unknown to you then leave the related fields at "SIP Parameters" blank. 2. Where can I get more info about the PSGw "Routing table"? I need more information regarding PSGw rules. Before starting using PSGw it is necessary to configure the "Routing table". The "Routing table" is a set of rules (3 in version 2.0) that defines handling of incoming Skype and SIP calls. Each rule has four parameters, namely: "Incoming username", "Incoming protocol", "Outgoing username", "Outgoing protocol". PSGw has some rules predefined, however it is necessary "to tune" them. 1. A very basic rule: Incoming username: * Incoming protocol: SIP Outgoing username: echo123 Outgoing protocol: Skype This rule defines that every incoming SIP call will be "routed" to the "echo123" (Skype Call Testing Service) Skype user. 2. A more complex rule: Incoming username: home Incoming protocol: SIP Outgoing username: iamattheoffice Outgoing protocol: Skype This rule defines that incoming SIP calls from the "home" user will be "routed" to the "iamattheoffice" Skype user. 3. Another basic rule: Incoming username: * Incoming protocol: Skype Outgoing username: 192.168.0.1 Outgoing protocol: SIP This rule defines that every incoming Skype call will be "routed" to the SIP endpoint with the 192.168.0.1 IP address. 4. Another complex rule: Incoming username: myboss Incoming protocol: Skype Outgoing username: me@192.168.0.1 Outgoing protocol: SIP This rule defines that calls from the "myboss" Skype user will be "routed" to the SIP client with following URI (Uniform Resource Locator) me@192.168.0.1 (see "SIP Basics" below). 3. SIP Basics Most common SIP addressing methods: - domain.com - host - name@domain.com - name@host - 192.168.0.1 (IP address) - name@192.168.0.1 (IP address) "SIP Proxy Server" is a server that receives SIP requests from a user agent or another proxy and acts on behalf of the user agent in forwarding requests or responding to them. A proxy server typically has access to a database or a location service to aid it in processing the request (determining the next hop). The interface between the proxy and the location service is not defined by the SIP protocol. The proxy can use any number of types of databases to aid in processing a request. Databases can contain SIP registrations, presence information, or any other type of information about the user location. The proxy server can also bypass media streams (audio, video and supplementary services). "SIP Redirect Server" is a type of SIP server that responds to requests, but does not forward them. Like a proxy sever, a redirect server uses a database or a location service to look up a user. The location information, however, is sent back to the caller in a redirection class response which finishes the transaction. "Pure" redirect servers are very rare. "SIP Registrar Server" accepts "registration" requests from the SIP endpoint and a client. Normally the contact information from the request is then made available to other SIP servers within the same administrative domain, such as proxies and redirect servers. Very often Registrar and Proxy works together at the same IP address. 4. Got "404 Error" at the SIP endpoint when placing a call from a SIP device to PSGw Some SIP endpoints can require the SIP address to be specified as "sip:192.168.0.1" instead of just "192.168.0.1". 5. OK, I can call "echo123". How can I connect to another Skype user? What should I specify as "Outgoing username"? To specify Skype address use Skype nicknames (also known as usernames). 6. Hmm, PSGw doesn't connect to the SIP gateway when I call PSGw from Skype. I see: "SIP/2.0 404 Not Found" in the SIP device log. This can happen if an IP address was specified as "Outgoing username". Try to use the following SIP addressing method "yourphonenumber@ip_address_of_SIP_device". Substitue "yourphonenumber" with your name at the SIP device. 7. Does PSGw relay or bypass DTMF tones? Currently PSGw doesn't support DTMF relaying. DTMF bypassing is allowed only from SIP/H.323 to Skype as there is no Skype API to recognize entered DTMF. Update: PSGw 2.5 will bypass DTMF from Skype to SIP/H.323. The release will be available soon. 8. Can I run PSGw in VMWare? It was reported that PSGw successfully run at Dual 3.2Ghz Pentium Xeon. Less CPU can potential reduce voice quality. 9. Can I run PSGw remotelly using Remote Desktop? It was reported that PSGw successfully runs in Remote Desktop session. Use it on your own risk or use VNC instead. 10. Are there any examples of SIP configuration for well-known SIP providers? Certainly. The following configurations that work with PSGw: VoIPBuster (One way to PSTN) SIP Parameters
SIP Parameters
No, this is not possible as this requires simulation of an incoming call to the local Skype client which is not possible due to the proprietary Skype protocol. 12. I run Skype and PSGw. Skype is logged in with the "myusername" Skype username. Can I use my local Skype to call SIP or H.323 phones? No, this is not possible as this requires simulation of an outgoing call from the local Skype client, which is not possible due to the proprietary Skype protocol. 13. I use PSGw with Asterisk. Sometimes Asterisk shows the "481 Call Leg/Transaction Does Not Exist" message. Why? This is OK. When PSGw registers Asterisk sends information about available voice mails (even if the mailbox is empty). As PSGw can't process voice mailbox indication it passes the error to Asterisk. This behavior doesn't affect PSGw or Asterisk performance. 14. I want to use PSGw with Asterisk. How should I configure PSGw? There are several ways to use PSGw with Asterisk. Here is the common way: First, configure "SIP Parameters": "Username": asterisk_username "Password": asterisk_password (if any) "Registrar address": asterisk_ip_address Next, configure "Routing table". 15. How must I configure Skype to use it with PSGw? Are there any specific requirements? Please ensure that the Skype "Auto answer" feature is switched off. As PSGw takes control over Skype please do not use Skype when PSGw is running. |
February 22, 2010. New version of H.323/SIP SoftPhone released February 21, 2010. New version of H.323/SIP MCU (Conference Server) released October 2, 2009. Press about us May 18, 2009. RSDevs.com is one of the Top 10 Finalists of Cisco Developer Contest !!! March 20, 2009. RSDevs.com introduces H.323/SIP Video Softphone for PDA - Windows Mobile 2003/5.0/6.0 Video SoftPhone for SIP and H.323 networks July 14, 2008. RSDevs.com introduces "Web-2-Phone" - Flash-based phone with SIP/H.323/Skype support July 19, 2007. PSGw 3.5 (SIP version) demo released February 1, 2007. PSGw for Linux 1.6 (SIP version) released January 1, 2007. Merry Christmas and Happy New Year! July 24, 2006. PSGw for Linux 1.5 (SIP version) released July 18, 2006. PSGw 3.5 Standard and Pro now has unique feature - arbitrary SIP destination! July 1, 2006. PSGw 3.3 Pro (SIP and H.323 version) allows to set SIP and H.323 DTMF type June 29, 2006. PSGw 3.3 Standard (SIP version) allows to set SIP DTMF type June 20, 2006. PSGw for Linux 1.2 (SIP version) is released for beta-testing! May 18, 2006. Updated PSGw has codec management feature and prefix for Skype username |