Re: [Asterisk-Users] codec negotiation
Hello Eduardo, Wednesday, December 17, 2003, 1:08:00 AM, you wrote: EG Hi list, EG I'm with a little problem on codec negotiation between a cisco827 and EG asterisk. EG My sip.conf is like that: EG [general] EG port = 5060 EG bindaddr = 0.0.0.0 EG context = default EG amaflags = default EG allow=g729 EG allow=gsm EG allow=alaw EG allow=ulaw EG ;disallow=all EG and cisco like that: EG dial-peer voice 6 voip EG destination-pattern 0T EG session protocol sipv2 EG session target ipv4:asterisk-ip EG dtmf-relay rtp-nte EG codec g711alaw EG no vad EG ! EG When I try to make a call, cisco shows codec g711alaw, but asterisk EG shows codec g729A (i have the licenses) and there is no audio. When I EG try disallow=g729, the same occurs, but this time asterisk shows codec EG gsm. EG The only way to make a call is allowing only alaw. But this is not EG convenience, since i need to use g279 with another endpoint (working EG ok). EG Why this negotiation problem happens? Try to add to cisco peer (not shown in your mail) [cisco] disallow=all allow=alaw -- Best regards, Nguyenmailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Where can i get the g.723 codec?
Hi Andrew, Tuesday, November 4, 2003, 5:04:04 AM, you wrote: AJ I have used G723.1 (although unlicensed) with Asterisk. The info is even AJ in the Makefile, just drop in a few files in your source directoy, AJ uncomment something in the Makefile and instant G723.1 support... Thanks for the info, but after looking at the Makefile, I am not sure where I can get the source code for G723? Can you provide a hint? TIA -- Best regards, Nguyenmailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] INFO method and DTMF translation
Hello guys,
I have searched high and low, but not found any information about
rules of using DTMF in SIP INFO method. Cisco has described something with
Signal=, but it look like this feature is dependent on implementors?
The problem is chan_sip.c cannot correctly translate received DTMF
digits, especially #,*. At least with my Antek EGW-804 gateway.
Looking into chan_sip.c, I found this code:
line 3982
if (p-owner) {
if (strlen(buf)) {
if (sipdebug)
ast_verbose(DTMF received: '%c'\n, buf[0]);
event = atoi(buf); WHY?
if (event 10) {
resp = '0' + event;
} else if (event 11) {
resp = '*';
} else if (event 12) {
resp = '#';
} else if (event 16) {
resp = 'A' + (event - 12);
}
memset(f, 0, sizeof(f));
f.frametype = AST_FRAME_DTMF;
f.subclass = resp;
f.offset = 0;
f.data = NULL;
f.datalen = 0;
ast_queue_frame(p-owner, f, 0);
}
On line 3986, any # or * digit I entered was translated to 0(zero). So
any apps depends on # for terminating (voicemail for example) won't
work.
My question is , why not take just buf[0]? why translate? my UA
always send something like d= (one digit) at a time.
--
Best regards,
Nguyen mailto:[EMAIL PROTECTED]
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Re[2]: [Asterisk-Users] INFO method and DTMF translation
Hello Brancaleoni, Sunday, October 12, 2003, 4:39:32 PM, you wrote: BM Hi. BM The implementation is correct, I can use sip info BM method to get all the DMTF, *,# included (eg voicemail BM works great with sip info dtmf) BM the line atoi(buf) is needed 'cause buf is a char, and BM we need a int value to do the comparisons below that line. BM and I don't see why they get set to 0 ... probably BM 'cause the INFO dtmf on your gw is broken. BM sip info method describes dmtf as numbers (not chars) , BM so 0-9 are the digits, 10,11 are respectively *,# BM and 12-16 are A,B,C,D BM if you get translated to 0, I can assume that your gw BM sends out # or * as char and not as numbers, as sip info BM method requires. BM matteo. Thanks matteo. You are correct. My gw send out # and * instead of 10 or 11. So I have created a 'special' handling for this broken?! implementation. -- Best regards, Nguyenmailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
