| Age | Commit message (Collapse) | Author | Files | Lines |
|---|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r282577 | dvossel | 2010-08-17 16:36:57 -0500 (Tue, 17 Aug 2010) | 16 lines
Merged revisions 282576 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010) | 9 lines
fixes no default transport for temp peer creation in chan_sip
(closes issue #17829)
Reported by: falves11
Patches:
issue_17829.rev1.txt uploaded by russell (license 2)
issue_17829.diff uploaded by dvossel (license 671)
Tested by: falves11
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282578 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines
remove current STUN support from chan_sip.c
This patch removes the current broken/useless stun
support from chan_sip.
(closes issue #17622)
Reported by: philipp2
Review: https://reviewboard.asterisk.org/r/855/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282304 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) | 4 lines
res_stun_monitor for monitoring network changes behind a NAT device
Review: https://reviewboard.asterisk.org/r/854
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282270 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r282236 | dvossel | 2010-08-13 13:58:10 -0500 (Fri, 13 Aug 2010) | 23 lines
Merged revisions 282235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010) | 16 lines
only do magic pickup when notifycid is enabled
A new way of doing BLF pickup was introduced into 1.6.2. This feature
adds a call-id value into the XML of a SIP_NOTIFY message sent to alert
a subscriber that a device is ringing. This option should only be enabled
when the new 'notifycid' option is set... but this was not the case. Instead
the call-id value was included for every RINGING Notify message, which
caused a regression for people who used other methods for call pickup.
(closes issue #17633)
Reported by: urosh
Patches:
chan_sip.txt uploaded by urosh (license )
blf_cid_issue.diff uploaded by dvossel (license 671)
Tested by: dvossel, urosh, okrief, alecdavis
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282237 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r281874 | mnicholson | 2010-08-11 16:11:54 -0500 (Wed, 11 Aug 2010) | 10 lines
handle all possible responses to REFER requests
(closes issue #17486)
Reported by: davidw
Patches:
Issue17486-counterbid.diff.txt uploaded by davidw (license 780)
Tested by: davidw
Review: https://reviewboard.asterisk.org/r/837/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281876 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r281760 | mnicholson | 2010-08-11 12:27:59 -0500 (Wed, 11 Aug 2010) | 4 lines
Avoid a deadlock in add_header_max_forwards().
Related to r276951
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281761 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r281532 | russell | 2010-08-10 11:54:20 -0500 (Tue, 10 Aug 2010) | 8 lines
Ensure that the proper external address is used for the RTP destination.
(closes issue #17044)
Reported by: ebroad
Tested by: ebroad
Review: https://reviewboard.asterisk.org/r/566/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281533 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r281432 | dvossel | 2010-08-09 15:47:53 -0500 (Mon, 09 Aug 2010) | 20 lines
Merged revisions 281430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010) | 13 lines
fixes SIP peers memory leak
We zeroed out the peer's addr before it was removed from the
peers_by_ip container. This made it impossible to be removed
from the container as the addr is the key used by the container
to find the peer.
(closes issue #17774)
Reported by: kkm
Patches:
017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281433 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r280778 | simon.perreault | 2010-08-03 15:54:03 -0400 (Tue, 03 Aug 2010) | 9 lines
Fixed IPv6-related SIP parsing bugs.
(closes issue #17663)
Reported by: oej
Patches:
diff uploaded by sperreault (license 252)
diff2 uploaded by sperreault (license 252)
get_domain.diff uploaded by sperreault (license 252)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280780 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
properly.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280745 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #17663)
Reported by: oej
Patches:
diff uploaded by sperreault (license 252)
diff2 uploaded by sperreault (license 252)
get_domain.diff uploaded by sperreault (license 252)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280707 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280627 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r280552 | dvossel | 2010-07-29 15:43:47 -0500 (Thu, 29 Jul 2010) | 17 lines
Merged revisions 280551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010) | 11 lines
fixes wrong SRV query for TLS connection
(closes issue #17612)
Reported by: marcelloceschia
Patches:
chan-sip_srvQuery.patch uploaded by marcelloceschia (license 1079)
chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia (license 1079)
Tested by: marcelloceschia, st, pabelanger
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280553 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280518 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280093 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r279887 | mmichelson | 2010-07-27 13:54:07 -0500 (Tue, 27 Jul 2010) | 16 lines
Fix parsing error in sip_sipredirect().
The code was written in a way that did a bad job of
parsing the port out of a URI. Specifically, it would
do badly when dealing with an IPv6 address. In this
particular scenario, there was no value from parsing
the port out, so I just removed that logic. And while
I was messing around in the function, I changed some
variable names to be more descriptive.
(closes issue #17661)
Reported by: oej
Patches:
17661.diff uploaded by mmichelson (license 60)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279888 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r279817 | dvossel | 2010-07-27 11:09:15 -0500 (Tue, 27 Jul 2010) | 2 lines
fix sip transaction match with authentication, fix confusing log message when using getaddrinfo
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279818 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r279785 | mmichelson | 2010-07-27 10:15:22 -0500 (Tue, 27 Jul 2010) | 20 lines
Merged revisions 279784 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul 2010) | 14 lines
Fix bad behavior of dynamic_exclude_static option in sip.conf.
We were attempting to create a contactdeny rule based on the peer's
IP address before the peer's IP address had been set. By moving the
processing further down in the function, we can ensure stuff works
as we expect for it to.
(closes issue #17717)
Reported by: mmichelson
Patches:
17717.patch uploaded by mmichelson (license 60)
Tested by: DennisD
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279786 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r279568 | dvossel | 2010-07-26 14:59:03 -0500 (Mon, 26 Jul 2010) | 21 lines
transaction matching using top most Via header
This patch modifies the way chan_sip.c does transaction to dialog
matching. Asterisk now stores information in the top most Via header
of the initial incoming request and compares that against other Requests
that have the same call-id. This results in Asterisk being able to
detect a forked call in which it has received multiple legs of the
fork. I completely stripped out the previous matching code and made
the comparisons a little more explicit and easier to understand. My
comments in the code should offer all the details involving this patch.
This patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to
find multiple dialogs with the same call-id. Since the callback
function was returning (CMP_MATCH | CMP_STOP) only the first item
found was being returned. I fixed this by making a new callback
function for finding multiple dialogs that only returns (CMP_MATCH)
on a match allowing for multiple items to be returned.
Review: https://reviewboard.asterisk.org/r/776/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279569 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This initially was created to work around the issue of
using a string comparison instead of a binary comparison
for IP addresses. It evolved a bit when test cases were
created and it was discovered that comparison of URI
parameters was not working exactly as it should.
sip_uri_cmp() and its helpers have been moved to reqresp_parser.c
and a new test has been added.
(closes issue #17662)
Reported by: oej
Review: https://reviewboard.asterisk.org/r/792
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278980 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278945 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278944 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Review: https://reviewboard.asterisk.org/r/795
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278908 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
If the Expire header of a SUBSCRIBE is less that our expiremin,
a log warning will be displayed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278619 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
min allowed
[RFC3265]3.1.6.1....
The notifier MAY also check that the duration in the "Expires" header
is not too small. If and only if the expiration interval is greater
than zero AND smaller than one hour AND less than a notifier-
configured minimum, the notifier MAY return a "423 Interval too
small" error which contains a "Min-Expires" header field. The "Min-
Expires" header field is described in SIP [1].
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278536 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278461 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
If Asterisk sends a 4xx error and the other side sends a CANCEl
before receiving the 4xx and responding with the ACK, Asterisk
will process the CANCEL and send a 487 Request Terminated as
a new final response to the INVITE. Since we are issuing a new
final response to the INVITE, the old one must be pretend_acked
else it will keep retransmitting.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278234 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
There are two changes here:
1. Since the externip setting can now have a port attached
to it, calling it "externip" is misleading. The option is now
documented and parsed as "externaddr." This also extends to the
"matchexterniplocally" setting. It is now documented and parsed
as "matchexternaddrlocally." The old names for the options may
still be used, but they are no longer used in the sip.conf.sample
file.
2. If no port is set for the externaddr, and UDP is the transport
to be used, then we will set the port of the externaddr to that of
the udpbindaddr. This was how things worked prior to the IPv6 merge,
so this is a regression fix.
(closes issue #17665)
Reported by: mmichelson
Patches:
17665.diff#2 uploaded by pprindeville (license 347)
Tested by: pprindeville
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277873 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
ACLs can now be configured to match IPv6 networks. This is only
relevant for ACLs in chan_sip for now since other channel drivers
do not support IPv6 addressing. However, once those channel drivers
are outfitted to support IPv6 addressing, the ACLs will already be
ready for IPv6 support.
https://reviewboard.asterisk.org/r/791
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277814 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul 2010) | 4 lines
Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested.
FAX-128
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277530 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277065 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Found a unused bag of curly brackets under my table. I always wondered where
they had gone. They where indeed needed in chan_sip.c
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276989 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
sip.conf configuration for the channel and for devices.
The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary,
like SIP proxys and SBCs, decrement this counter and detects when it reaches zero,
at which point the SIP request is nicely killed in a SIP-friendly way.
Review: https://reviewboard.asterisk.org/r/778/
Thanks to dvossel for the review and good advice.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276951 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Found based on message from Philip Prindeville on the
Asterisk Developers mailing list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276909 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Related to changes from 276571, I was accidentally testing with a port set in
my configuration causing me to miss this. Also moved the TCP handling as well
to occur before build_peer is called.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276788 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
MWI updates were not being sent if no messages were found in the event cache.
This was corrected since a phone may need to clear its MWI status configured
previously from another mailbox.
Upon module or sip reload, MWI updates could not be sent due to the sipsock
socket not being set early enough in reload_config. The code handling the
descriptor assignment and such has simply been moved before the call to
build_peer.
Issuing a sip reload cleared the IP address of the peer, but skipped checking
the database for registration information. The database is now checked both
for sip reload and actually reloading the module.
If a transmission occurs before the do_monitor thread has started, do not
attempt to send a signal to it.
(closes issue #17398)
Reported by: ip-rob
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276571 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
ast_sockaddr_stringiy_fmt (which is call by all ast_sockaddr_stringify* functions)
uses thread-local storage for storing the string that it creates. In cases where
ast_sockaddr_stringify_fmt was being called twice within the same statement, the
result of one call would be overwritten by the result of the other call. This
usually was happening in printf-like statements and was resulting in the same
stringified addressed being printed twice instead of two separate addresses.
I have fixed this by using ast_strdupa on the result of stringify functions if
they are used twice within the same statement. As far as I could tell, there were
no instances where a pointer to the result of such a call were saved anywhere, so
this is the only situation I could see where this error could occur.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276570 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Unlike most responses, the 200 Ok to a pending INVITE Request is
acknowledged by an ACK Request. If the ACK Request for this Response is not received
the previous behavior was to immediately destroy the dialog and hangup
the channel. Now in an effort to be more RFC compliant, instead of immediately
destroying the dialog during this special case, termination is done with a BYE Request
as the dialog is technically confirmed when the 200 Ok is sent even if the ACK is
never received. The behavior of immediately hanging up the channel remains.
This only affects how dialog termination proceeds for this one special case.
RFC 3261 section 13.3.1.4
"If the server retransmits the 2xx response for 64*T1 seconds without receiving
an ACK, the dialog is confirmed, but the session SHOULD be terminated. This is
accomplished with a BYE, as described in Section 15."
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276439 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.
This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/
Review: https://reviewboard.asterisk.org/r/744/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276393 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
I've been working in this function a bunch lately, and
these huge debug strings are getting annoying.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276392 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
merge mistake from IPv6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276389 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Retransmission of packets should not be based on how many packets were
sent, but instead on a timeout period. Depending on whether or not the
packet is for a INVITE or NON-INVITE transaction, the number of packets
sent during the retransmission timeout period will be different, so
timing out based on the number of packets sent is not accurate.
This patch fixes this by removing the retransmit limit and only stopping
retransmission after a timeout period is reached. By default this
timeout period is 64*(Timer T1) for both INVITE and non-INVITE
transactions. For more information on sip timer values refer to
RFC3261 Appendix A.
Review: https://reviewboard.asterisk.org/r/749/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276219 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Some code improperly assumes that the sessions are still there, so revert the
change until I can find all of them and fix them.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276206 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Since we are only keeping the dialog around for retransmissions at this point
and there is no possibility that we are still handling RTP, go ahead and
destroy the RTP sessions. Keeping them alive for 32 past when they are used
is unnecessary and can lead to problems with having too many open file
descriptors, etc.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275998 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275816 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
It was essentially an off-by-one error. The easiest way
to fix this was to use the handy-dandy AST_NONSTANDARD_RAW_ARGS
macro to parse the pieces of the registration string out. Tested
and it works wonderfully.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275385 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
If a Via header contained an IPv6 address, we would not properly parse
the port. We would instead get the information after the first colon in
the address.
(closes issue #17614)
Reported by: oej
Patches:
diff uploaded by sperreault (license 252)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275308 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This should fix all the CDR tests that were not passing. When they would
originate a call, all fields in the INVITE that contained the source port would
have the port set to 0. Most troubling of these was the Contact header. Tests
are passing locally now and should also pass on the bamboo build agents.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275294 f38db490-d61c-443f-a65b-d21fe96a405b
|
