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2010-10-19Add sip show peer info about crypto and remove dated commenttwilson1-0/+2
This patch adds information about the encryption setting to 'sip show peers' and removes an out-of-date comment from res_srtp.c and instead directs users to the proper documentation. (closes issue #18140) Reported by: chodorenko git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292309 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-15Fixes peer's host port information being lost on sip reload.dvossel1-0/+3
(closes issue #18135) Reported by: lmadsen Patches: crazy_ports_v2.diff uploaded by dvossel (license 671) Tested by: lmadsen git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291942 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-14Add the ability for ast_find_ourip to return IPv4, IPv6 or both.pabelanger1-1/+1
While testing chan_gtalk I noticed jabber was using my IPv6 address and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip() to return both IPv6 and IPv4 results. Adding a family parameter gives you the ablility to choose. Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results. Review: https://reviewboard.asterisk.org/r/973/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291758 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revisions 291393 via svnmerge from russell1-0/+16
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291393 | russell | 2010-10-13 10:29:21 -0500 (Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines Lock pvt so pvt->owner can't disappear when queueing up a frame. This fixes a crash due to a hangup race condition. ABE-2601 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291394 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Move declaration closer to where now used.rmudgett1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291113 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Merged revisions 291110-291111 via svnmerge from rmudgett1-4/+9
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500 (Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) | 1 line Add missing unlock to an exception condition in reload_config(). ........ ................ r291111 | rmudgett | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit from handle_request_do() consistent. ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291112 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-02Merged revisions 289798 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289840 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Merged revisions 289700 via svnmerge from jpeeler1-3/+8
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500 (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) | 14 lines Ensure user portion of SIP URI matches dialplan when using encoded characters. This commit takes a simliar approach to 288112 and checks the dialplan to determine the proper action for an incoming contact header as to whether or not it should be decoded or not. sip_new was blindly always decoding the extension, which also caused the outgoing contact header to be incorrect as well as failing to match the encoded extension in the dialplan. (closes issue #17892) Reported by: wdoekes Patches: bug17892-1.patch uploaded by jpeeler (license 325) Tested by: wdoekes ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289701 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01don't iterate through all dialogs to find and delete old subscribesschmitds1-36/+5
On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed. (closes issue #17950) Reported by: schmidts Tested by: schmidts Review: https://reviewboard.asterisk.org/r/901/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289622 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-30Merged revisions 289553 via svnmerge from mnicholson1-2/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep 2010) | 4 lines Properly handle channel allocation failures duing invites with replaces. ABE-2588 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289554 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-28Break up long ast_manager_event_multichan() event lines.rmudgett1-3/+28
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289054 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-27Still build SIP, even if res_crypto cannot be built (use, not depend).tilghman1-1/+1
(closes issue #18062) Reported by: a user on the mailing list git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@288961 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-24Append Retry-After header on 500 error response to Re-INVITE according to ↵dvossel1-2/+17
RFC3261 section 14.2. ABE-2301 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@288852 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-24Inspect Require header on BYE transaction according to RFC3261 section 8.2.2.3.dvossel1-2/+19
ABE-2293 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@288821 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Merged revisions 288417 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500 (Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) | 5 lines RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response. ABE-2458 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@288418 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Merged revisions 288344 via svnmerge from dvossel1-2/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500 (Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010) | 2 lines During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@288345 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Merged revisions 288113 via svnmerge from tilghman1-9/+18
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500 (Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) | 15 lines Try both the encoded and unencoded subscription URI for a match in hints. When a phone sends an encoded URI for a subscription, the URI is not matched with the actual hint that is in decoded format. For example, if we have an extension with a hint that is named: "#5601" or "*5601", the subscription will work fine if the phone subscribes with an already decoded URI, but when it's decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the correct hint. (closes issue #17785) Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@288159 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Send a "415 Unsupported Media Type" after failure to process sdp due to ↵dvossel1-3/+17
unknown Content-Encoding header. ABE-2258 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@287929 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Don't use ast_strdupa() from within the arguments to a function.russell1-13/+19
(closes issue #17902) Reported by: afried Patches: issue_17902.rev1.txt uploaded by russell (license 2) Tested by: russell Review: https://reviewboard.asterisk.org/r/927/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@287895 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Anonymous callerid needs a "sip:" uri prefix.tilghman1-2/+2
(closes issue #17981) Reported by: avalentin Patches: sip-anonymous-aastra.patch uploaded by avalentin (license 1107) (plus an additional fix by me) Tested by: avalentin git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@287893 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-20Fixes issue with registrations not working properly with pedantic=yes.dvossel1-2/+4
(closes issue #18017) Reported by: schmidts Patches: issues_18017_v1.diff uploaded by dvossel (license 671) Tested by: schmidts git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@287645 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-15Add parking extension for non-default parking lots.jpeeler1-9/+16
This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@286931 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-15Set tohost to the domain specified in the configuration file instead of the ↵mnicholson1-3/+2
IP address of the host we are calling. This fixes a regression introduced in r274783. (closes issue #17960) Reported by: adriavidal Patches: sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested by: mich, mnicholson, adriavidal (closes issue #17676) Reported by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@286868 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Sets subscribed type for outgoing MWI subscriptions so correct Event header ↵dvossel1-1/+3
is used. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@286834 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Merged revisions 286757 via svnmerge from mnicholson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500 (Tue, 14 Sep 2010) | 20 lines Merged revisions 286756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) | 13 lines Don't clear the username from a realtime database when a registration expires. Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either. (closes issue #17551) Reported by: ricardolandim Patches: reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96) Tested by: ricardolandim, mnicholson ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@286758 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-13Merged revisions 286456 via svnmerge from qwell1-1/+0
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) | 5 lines Remove "Internal IP" from sip show settings, as it's not at all useful to display. (closes issue #17840) Reported by: oej ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@286457 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Merged revisions 285710 via svnmerge from bbryant1-232/+374
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285711 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-03Fixed IPv6-related SIP parsing bugs.simon.perreault1-12/+34
(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/branches/1.8@280778 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Merged revisions 280551 via svnmerge from dvossel1-4/+33
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/branches/1.8@280552 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Fix parsing error in sip_sipredirect().mmichelson1-21/+16
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/branches/1.8@279887 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27fix sip transaction match with authentication, fix confusing log message ↵dvossel1-3/+2
when using getaddrinfo git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279817 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Merged revisions 279784 via svnmerge from mmichelson1-8/+9
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/branches/1.8@279785 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-26transaction matching using top most Via headerdvossel1-42/+232
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/branches/1.8@279568 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23SIP URI comparison fixes.mmichelson1-286/+8
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
2010-07-23... just kidding. Enable SIP by default. :-)russell1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278945 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Disable SIP support by default for Asterisk 1.8.russell1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278944 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Allow IPv6 addresses for UDPTL streams.mmichelson1-47/+26
Review: https://reviewboard.asterisk.org/r/795 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278908 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-22update sip subscription debug message to a warning messagedvossel1-2/+4
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
2010-07-21send "423 Interval too small" Response to Subscribe with Expires less that ↵dvossel1-2/+19
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
2010-07-21Properly set the port number for UDPTL media sessions.mnicholson1-2/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278461 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20fixes sip CANCEL race conditiondvossel1-10/+11
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
2010-07-20Add load priority order, such that preload becomes unnecessary in most casestilghman1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-19Fix port setting of external address in SIP.mmichelson1-41/+43
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
2010-07-19Make ACLs IPv6-capable.mmichelson1-50/+25
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
2010-07-16Merged revisions 277497 via svnmerge from mnicholson1-0/+4
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
2010-07-16Formatting fixesoej1-168/+306
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277065 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Formatting changes (guideline corrections)oej1-35/+69
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
2010-07-16Add ability to configure the Max-Forwards header in the dialplan, as well as inoej1-3/+45
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
2010-07-16Fix reversed logic of if statement.mmichelson1-1/+1
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
2010-07-15Correct not setting the bindport before attempting to open the socket.jpeeler1-64/+62
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
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-27 22:55:08 +0000