[asterisk-users] inbound sip rtcp hangup

2006-07-19 Thread Vincent Regnard
Hi all, I have configured a connection my sip voip provider. I can make outbound call without trouble. But I cannot recieve voip calls. The sip negociation seams to start well but at some point during the rtcp dialog, things seems to block. As you can see on the above log sample, I recieve

[asterisk-users] SIP_HEADER() read-only

2006-08-02 Thread Vincent Regnard
Hi, Having checked the documentation for SIP_HEADER: pitux-exercice15*CLI -= Info about function 'SIP_HEADER' =- [Syntax] SIP_HEADER(name) [Synopsis] Gets or sets the specified SIP header I thought I could write some info in SIP_HEADER to retrieve them later. But when I try to write to

Re: [asterisk-users] SIP_HEADER() read-only

2006-08-03 Thread Vincent Regnard
There is presently no .write member in the structure declaration for this function in channels/app_sip.c: static struct ast_custom_function sip_header_function = { .name = SIP_HEADER, .synopsis = Gets or sets the specified SIP header, .syntax = SIP_HEADER(name),

Re: [asterisk-users] SIP_HEADER() read-only

2006-08-04 Thread Vincent Regnard
Joshua Colp a écrit : You can use the SIPAddHeader application: SIPAddHeader(Header: Content) Adds a header to a SIP call placed with DIAL. Remember to user the X-header if you are adding non-standard SIP headers, like X-Asterisk-Accountcode:. Use this with care. Adding the wrong headers may