[asterisk-dev] DTMF offer oddities

Hi, I've noticed some changes in handling of DTMF offers in SIP packages since my last update of Asterisk to 1.2.16. The behavior was somehow "foreseeable" still at version 1.2.13, at 1.2.15 it was already "different". Don't know about 1.2.14. Here is how it should be (according to my knowledge)

[asterisk-dev] Asterisk 1.4.x compilation problem

Hi, all, I've been using Asterisk 1.2. And having problems to update to Asterisk 1.4. Both 1.4.0 and 1.4.1 does not compile on my gentoo box with GCC 4.0 with OpenLine4 Card (vpb). In 1.4.0, in channels make compiled chan_vpb.cc into chan_vpb.o and chan_vpb.oo, by removing one of the object f

[asterisk-dev] please tell me

Hello , i installed astrisk 1.4.1 , but i want to add user in sip.conf file and number in extension.conf file . but there is no such file given . can anybody tell me , what to do ? regards mehul ___ --Bandwidth and Colocation provid

[asterisk-dev] Asterisk 1.2.17 Released

The Asterisk and Zaptel development teams have released Asterisk version 1.2.17. Along with minor bug fixes, this release incorporates a fix for the SIP DoS vulnerability recently discovered by INRIA Lorraine (http://voipsa.org/pipermail/voipsec_voipsa.org/2007-March/002275.html). All users of As

Re: [asterisk-dev] please tell me

Hello Mehul, This is not the correct mailing list for this type of question. If you post this message to the asterisk-users list, you are much more likely to receive a helpful response. Good luck, Sean On 3/21/07, mehul shah <[EMAIL PROTECTED]> wrote: Hello , i installed ast

[asterisk-dev] Asterisk 1.4.2 Released

The Asterisk and Zaptel development teams have released Asterisk 1.4.2. In addition to minor bug fixes, this release includes: - improved SLA support, sample configurations and documentation - fixes for incoming DTMF handling in the IAX2 channel driver There are also two security-related change

[asterisk-dev] playback--> urgent issue

 Hello all, Rightnow i am using AMI with socket based interface, it will act as a server. The client will send a command, in this command contains FILENAME and CHANNELID of already established call . so, i can receive this command via socket, after that i will play the given file name for men

[asterisk-dev] Possible deadlock situation in chan_sip.c?

Hi! Can this code section from chan_sip.c (begin at line 14920) cause deadlocks if any error occurs between the mutex lock for iflock? --- snip --- /* Check for interfaces needing to be killed */ ast_mutex_lock(&iflock); restartsearch: t = time(NULL);

Re: [asterisk-dev] please tell me

Please contact the asterisk-users mailing list. this list is for dev only. On 3/21/07, mehul shah <[EMAIL PROTECTED]> wrote: Hello , i installed astrisk 1.4.1 , but i want to add user in sip.conf file and number in extension.conf file . but there is no such file given . can a

[asterisk-dev] Request for new MIME type in asterisk webserver

Hi all, Can some one please add the following mime type in asterisk webserver. http.c --- mimetypes[] = { { "png", "image/png" }, { "jpg", "image/jpeg" }, { "js", "application/x-javascript" }, { "wav", "audio/x-wav" }, { "mp3", "audio/mpeg" }, +