[SR-Users] ISUP decode and store to be able to append_body_part() later

2015-07-15 Thread andre second
Hi, I am trying to extract and save ISUP from SIP-I packet for a later use. The problem is that data is stored in binary format and needs decoding before saving to a variable. Can Kamailio perform any decoding so that later I can use append_body_part() to append it back to a multipart message?

Re: [SR-Users] Textops and Multipart Body - adding ISUP segfaults Kamailio

2015-07-08 Thread andre second
Hi Victor, Many thanks for your reply. Here is what happens if I amend msg_apply_changes():  /usr/sbin/kamailio[1705]: ERROR: textops [textops.c:1854]: append_multibody_helper(): Cannot get boundary. Is body multipart?  /usr/sbin/kamailio[1705]: INFO:

[SR-Users] Textops and Multipart Body - adding ISUP segfaults Kamailio

2015-07-07 Thread andre second
Hi, I am using Kamailio version: kamailio 4.3.0 (x86_64/linux) c6aa95 on CentOS 6 I am trying to encapsulate ISUP in the INVITE: ...if(has_body("application/sdp")){      set_body_multipart();         if(msg_apply_changes())         {              $var(acm) = "7e Od 04 55 75 69 20 4d 61 6b 65 43 61

Re: [SR-Users] From sip phone to provider trunk

2012-12-17 Thread andre second
Still no success.Shall I use auth module maybe? --- On Fri, 12/14/12, andre second wrote: From: andre second Subject: Re: [SR-Users] From sip phone to provider trunk To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" , mico...@gmail.com Da

Re: [SR-Users] From sip phone to provider trunk

2012-12-14 Thread andre second
outer (SER) - Users Mailing List" Cc: "andre second" Date: Thursday, December 13, 2012, 11:14 AM Hello, is your trunk provider requiring a username/password for the calls sent to it, or it is just IP based peering? Cheers, Daniel

[SR-Users] From sip phone to provider trunk

2012-12-10 Thread andre second
Hi,I have some sip phones and using them to register at Kamailio which is located behind 2 asterisk servers. There 2 SIP trunks to my SIP provider on asterisk machines. Also I have rtpproxy running. What I want to do is to put some of the calls directly from the phones to SIP Provider without in