Re: [Asterisk-Users] Call Waiting on SIP phones
Subject: Re: [Asterisk-Users] Call Waiting on SIP phones Paul, I applied the patch successfully last night on the CVS from last night. I set the incominglimit=1 in sip.conf. I am still getting the ring in the Grandstream phones. I posted a bug report on the bugtracker. If you would like more system information, please let me know. thanks, Walker -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Waiting on SIP phones
- Original Message - From: Walker Haddock [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 22, 2003 11:10 PM Subject: Re: [Asterisk-Users] Call Waiting on SIP phones Subject: Re: [Asterisk-Users] Call Waiting on SIP phones Paul, I applied the patch successfully last night on the CVS from last night. I set the incominglimit=1 in sip.conf. I am still getting the ring in the Grandstream phones. I posted a bug report on the bugtracker. If you would like more system information, please let me know. thanks, Walker Hi Walker, I tried to duplicate your problem, but I couln't, could you please let me know the exact calling situation, so I could try and duplicate it. Also noted that you have a repeat SIP number in your Dial command for extension 203 (don't think that could impact it). Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Waiting on SIP phones
On Tue, Oct 21, 2003 at 09:32:44AM +1000, Paul Liew wrote:
Sorry, to repost - but I left a /* comment - here it is again
Paul
--- chan_sip.c.save 2003-10-20 21:51:52.0 +1000
+++ chan_sip.c 2003-10-21 09:26:41.0 +1000
@@ -959,7 +959,9 @@
return 0;
}
switch(event) {
+ /* Incoming and outging affects the inUse counter */
case DEC_IN_USE:
+ case DEC_OUT_USE:
if ( u-inUse 0 ) {
u-inUse--;
} else {
@@ -967,6 +969,7 @@
}
break;
case INC_IN_USE:
+ case INC_OUT_USE:
if (u-incominglimit 0 ) {
if (u-inUse = u-incominglimit) {
ast_log(LOG_ERROR, Call from user '%s'
rejected due to usage limit of %d\n, u-name, u-incominglimit);
@@ -977,6 +980,8 @@
u-inUse++;
ast_log(LOG_DEBUG, Call from user '%s' is %d out of %d\
n, u-name, u-inUse, u-incominglimit);
break;
+ /* Commented out - don't want to limit outgoing */
+ /*
case DEC_OUT_USE:
if ( u-outUse 0 ) {
u-outUse--;
@@ -994,6 +999,7 @@
}
u-outUse++;
break;
+ */
default:
ast_log(LOG_ERROR, find_user(%s,%d) called with no even
t!\n,u-name,event);
}
@@ -1086,6 +1092,12 @@
INVITE, but do set an autodestruct just in ca
se. */
needdestroy = 0;
sip_scheddestroy(p, 15000);
+ /* channel still up - reverse dec of inuse count
er */
+ if ( p-outgoing ) {
+ find_user(p, INC_OUT_USE);
+ } else {
+ find_user(p, INC_IN_USE);
+ }
} else {
char *res;
if (ast-hangupcause ((res = hangup_cause2sip
(ast-hangupcause {
Paul, I'm getting a patch error when I diff to the chan_sip.c that I just got from CVS
this morning. It looks like this morning's version hasn't changed from the version I
had from 9/24/03. Here's the .rej file output:
***
*** 1071,1076
INVITE, but do set an autodestruct just in case. */
needdestroy = 0;
sip_scheddestroy(p, 15000);
} else {
char *res;
if (ast-hangupcause ((res =
hangup_cause2sip(ast-hangupcause {
--- 1080,1091
INVITE, but do set an autodestruct just in case. */
needdestroy = 0;
sip_scheddestroy(p, 15000);
+ /* channel still up - reverse dec of inuse counter */
+ if ( p-outgoing ) {
+ find_user(p, INC_OUT_USE);
+ } else {
+ find_user(p, INC_IN_USE);
+ }
} else {
char *res;
if (ast-hangupcause ((res =
hangup_cause2sip(ast-hangupcause {
Here's what I find in the source around those lines:
needdestroy = 1;
/* Start the process if it's not already started */
if (!p-alreadygone strlen(p-initreq.data)) {
if (needcancel) {
if (p-outgoing) {
transmit_request_with_auth(p, CANCEL, p-ocseq, 1);
/* Actually don't destroy us yet, wait for the 487 on
our original
INVITE, but do set an autodestruct just in case. */
needdestroy = 0;
sip_scheddestroy(p, 15000);
} else
transmit_response_reliable(p, 403 Forbidden,
p-initreq);
} else {
if (!p-pendinginvite) {
/* Send a hangup */
transmit_request_with_auth(p, BYE, 0, 1);
} else {
/* Note we will need a BYE when this all settles out
but we can't send one while
Re: [Asterisk-Users] Call Waiting on SIP phones
- Original Message - From: Walker Haddock [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 22, 2003 5:50 AM Subject: Re: [Asterisk-Users] Call Waiting on SIP phones Paul, I'm getting a patch error when I diff to the chan_sip.c that I just got from CVS this morning. It looks like this morning's version hasn't changed from the version I had from 9/24/03. Here's the .rej file output: Walker, in case I did something wrong while posting the patch on to the list - try the patch I've put up on bugtracker http://bugs.digium.com/bug_view_page.php?bug_id=408 I've applied that patch on to the latest cvs and its ok. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Waiting on SIP phones
On Wed, Oct 22, 2003 at 09:46:52AM +1000, Paul Liew wrote: - Original Message - From: Walker Haddock [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 22, 2003 5:50 AM Subject: Re: [Asterisk-Users] Call Waiting on SIP phones Paul, I'm getting a patch error when I diff to the chan_sip.c that I just got from CVS this morning. It looks like this morning's version hasn't changed from the version I had from 9/24/03. Here's the .rej file output: Walker, in case I did something wrong while posting the patch on to the list - try the patch I've put up on bugtracker http://bugs.digium.com/bug_view_page.php?bug_id=408 I'm sorry I didn't make it clear. I did download the patch from the bug tracker. I've applied that patch on to the latest cvs and its ok. I have edited your code into the chan_sip.c source and compiled. I am testing. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Waiting on SIP phones
Sorry, to repost - but I left a /* comment - here it is again Paul [code block removed] Paul - A few questions and comments: 1) So, does this also make incominglimit and outgoinglimit work as expected? The current method doesn't do quite what the average user thinks it would do. 2) Your patch may be relevant to: http://bugs.digium.com/bug_view_page.php?bug_id=329 - I didn't realize that outgoinglimit=1 would only work for the first call, but fail subsequently. With type=peer, the incoming= and outgoing= modifiers really don't work at all for me, which makes them almost completely useless. If you have some method to fix that: great! 3) If there is an existing bugtracker item, the source code diff's are best appended to that particular bug. If there isn't one open, go ahead and open one. Sending diff's to the mailing list is getting less common (and less desirable) as time goes on, since we have the bugtracker now. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Waiting on SIP phones
On Monday 20 October 2003 18:21, Paul Liew wrote: Hi All, This is the first time I'm submitting a patch, and I hope it fixes more than it breaks. I'm putting it here, since John Todd mentioned a while ago about the heavy load Mark and crew have at Digium (doing such good work), so I thought all of us could test this first, and if ok submit for inclusion in CVS later if appropriate. snip You may want to do what everybody else does and open a bug report with this patch uploaded as a separate file to bugs.digium.com. Prefix the title of the patch with [patch] to denote it as more than just a problem report. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Waiting on SIP phones
- Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 21, 2003 10:29 AM Subject: Re: [Asterisk-Users] Call Waiting on SIP phones Paul - A few questions and comments: 1) So, does this also make incominglimit and outgoinglimit work as expected? The current method doesn't do quite what the average user thinks it would do. Yes it does John - however, I've made incominglimit and outgoinglimit to imply the same, so we only need incominglimit=1. 2) Your patch may be relevant to: http://bugs.digium.com/bug_view_page.php?bug_id=329 - I didn't realize that outgoinglimit=1 would only work for the first call, but fail subsequently. With type=peer, the incoming= and outgoing= modifiers really don't work at all for me, which makes them almost completely useless. If you have some method to fix that: great! The work once applies to incoming (never tested for outgoing) - Its now fixed for both - so let me know how you go. 3) If there is an existing bugtracker item, the source code diff's are best appended to that particular bug. If there isn't one open, go ahead and open one. Sending diff's to the mailing list is getting less common (and less desirable) as time goes on, since we have the bugtracker now. I've added to the bugtracker - http://bugs.digium.com/bug_view_page.php?bug_id=408 I've made it new report as it also fixes the incoming call waiting which 329 didn't refer to. Also thanks for your comments and also from Tilghman. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
