October 2010 Archives by subject

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Starting: Fri Oct 1 06:22:07 UTC 2010
Ending: Fri Oct 29 17:36:16 UTC 2010
Messages: 79

• [SIPForum-discussion] About RTP/RTCP   andre heller
• [SIPForum-discussion] Analyse SIP logs   Amruth yakkundi
• [SIPForum-discussion] Analyse SIP logs   Underwood, Rosa
• [SIPForum-discussion] Analyse SIP logs   Nitin Kapoor
• [SIPForum-discussion] Analyse SIP logs   Nitin Kapoor
• [SIPForum-discussion] Beep tones with SIP   Guy Churchouse
• [SIPForum-discussion] Branch and Tag parameters in SIP method   Nittin Dutt
• [SIPForum-discussion] c-seq value in SIP call   murali shankar
• [SIPForum-discussion] c-seq value in SIP call   Nitin Kapoor
• [SIPForum-discussion] c-seq value in SIP call   Ruben Roque
• [SIPForum-discussion] c-seq value in SIP call   Ruben Roque
• [SIPForum-discussion] c-seq value in SIP call   Ronald del Rosario
• [SIPForum-discussion] Call Failed with SIP 486 Busy for G.723 codec   Fatine Taoufik
• [SIPForum-discussion] Call Failed with SIP 486 Busy for G.723 codec   AMIT ANAND
• [SIPForum-discussion] Call Failed with SIP 486 Busy for G.723 codec   Sam
• [SIPForum-discussion] Call Failed with SIP 486 Busy for G.723 codec   Nitin Kapoor
• [SIPForum-discussion] Call Failed with SIP 486 Busy for G.723 codec   Kesavan Ramasamy
• [SIPForum-discussion] Call Failed with SIP 486 Busy for G.723 codec   amit salunkhe
• [SIPForum-discussion] Can you tell me how to implement dial plan   孙永光
• [SIPForum-discussion] Can you tell me how to implement dial plan   Donald Lee
• [SIPForum-discussion] Connected Number   Jyoti Parija
• [SIPForum-discussion] Connected Number   Caglar ER
• [SIPForum-discussion] definition for the terms   murali shankar
• [SIPForum-discussion] definition for the terms   Vijayalakshmi
• [SIPForum-discussion] DNS Name encoding in Response   Ashutosh Panda
• [SIPForum-discussion] FAX Issue in VoIP   murali shankar
• [SIPForum-discussion] FAX Issue in VoIP   murali shankar
• [SIPForum-discussion] FAX Issue in VoIP   Maciej Wasiel
• [SIPForum-discussion] FAX Issue in VoIP   shawn.degraw at comcast.net
• [SIPForum-discussion] FAX Issue in VoIP   Deepak Baskaran
• [SIPForum-discussion] FAX Issue in VoIP   Sam
• [SIPForum-discussion] FAX Issue in VoIP   kamalraj raj
• [SIPForum-discussion] FAX Issue in VoIP   Karthik Ramiya Rengarao (kramiyar)
• [SIPForum-discussion] Headers modified by B2BUA on receiving an INVITE request from a UAC   Ruben Roque
• [SIPForum-discussion] Headers modified by B2BUA on receiving an INVITE request from a UAC   AMIT ANAND
• [SIPForum-discussion] Implementation of RTP   andre heller
• [SIPForum-discussion] Is TCP feasible for SIP??   Vijayalakshmi
• [SIPForum-discussion] Is TCP feasible for SIP??   Maciej Wasiel
• [SIPForum-discussion] Is TCP feasible for SIP??   Sam
• [SIPForum-discussion] Is TCP feasible for SIP??   Medve, Steve
• [SIPForum-discussion] Is TCP feasible for SIP??   M. Ranganathan
• [SIPForum-discussion] Need help with particular my own scenario- Automated Testing   Guddu Dhanoa
• [SIPForum-discussion] number no geographic USA 855   Gomez Leopoldo
• [SIPForum-discussion] OCS to PSTN Calling stopped working   Thistle,Jeff
• [SIPForum-discussion] OCS to PSTN Calling stopped working   Sam
• [SIPForum-discussion] pjsip port with Mosync sdk........   satish gupta
• [SIPForum-discussion] Reg : Issue with 40 min call disconnection   Sathya
• [SIPForum-discussion] Reg : Issue with 40 min call disconnection   Sathya
• [SIPForum-discussion] Reg : Issue with 40 min call disconnection   vidyut gupta
• [SIPForum-discussion] Reg : Issue with 40 min call disconnection   Banda, Srinivas
• [SIPForum-discussion] Regarding "t=" line SDP   praveena ss
• [SIPForum-discussion] RTP generator and QoS tools needed   Raghavendra GL
• [SIPForum-discussion] RTP generator and QoS tools needed   Soumit
• [SIPForum-discussion] RTP implementation   andre heller
• [SIPForum-discussion] RTP implementation   andre heller
• [SIPForum-discussion] RTP implementation   Maciej Wasiel
• [SIPForum-discussion] RTT estimation in SIP   mustafa rifaee
• [SIPForum-discussion] RTT estimation in SIP   mustafa rifaee
• [SIPForum-discussion] RTT estimation in SIP   Sumeetkumar Bhardwaj
• [SIPForum-discussion] SIP based Middleware   Hakiri Akram
• [SIPForum-discussion] SIP based Middleware   Nitin Kapoor
• [SIPForum-discussion] SIP Client-Server interaction!   andre heller
• [SIPForum-discussion] SIP Client-Server interaction!   raj kumaradass
• [SIPForum-discussion] SIP Client-Server interaction!   andre heller
• [SIPForum-discussion] SIP Forum’s SIPit 27 Slated for Taipei, Taiwan, November 15-19, 2010   Marc Robins
• [SIPForum-discussion] SIP REFER and new INVITE   sampath.kumar1 at wipro.com
• [SIPForum-discussion] SIP REFER and new INVITE   praveena ss
• [SIPForum-discussion] SIP REFER and new INVITE   sampath.kumar1 at wipro.com
• [SIPForum-discussion] SIP signaling based Middleware   Hakiri Akram
• [SIPForum-discussion] SIP Video Calls & PSTN/GSM Interface   shaikh. bashaa
• [SIPForum-discussion] SIPit registration closes in one week   Robert Sparks
• [SIPForum-discussion] SIPit27 registration closes in 3 weeks   Robert Sparks
• [SIPForum-discussion] Smart Grids Data Management Event Details   Kim Schofield
• [SIPForum-discussion] Smart Grids Data Management Event Details   Kim Schofield
• [SIPForum-discussion] Soft phone for Linux OS   samuel
• [SIPForum-discussion] URI in the TO header   FRANCISCO GIOIA
• [SIPForum-discussion] URI in the TO header   shawn.degraw at comcast.net
• [SIPForum-discussion] Use of C-seq   sanjay Kumar
• [SIPForum-discussion] Where should INVITE w/o SDP needs to be sent in call transfer scenario   mustafa aydin

Last message date: Fri Oct 29 17:36:16 UTC 2010
Archived on: Sat Apr 22 08:35:24 UTC 2017

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