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git:rbetancor/drouting: drouting: Documentation updated

by Raul Alexis Betancor Santana

Module: sip-router Branch: rbetancor/drouting Commit: 9d8cb1917800e096d8a34232e63fb01ca83855d7 URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=9d8cb19… Author: Raul Alexis Betancor Santana <rabs(a)dimension-virtual.com> Committer: Raul Alexis Betancor Santana <rabs(a)dimension-virtual.com> Date: Sat Jun 20 00:58:32 2009 +0100 drouting: Documentation updated - entities.xml path fixed - &osips references changed to &kamailio references - README regenated --- modules_k/drouting/README | 825 ++++++++++++++++------------- modules_k/drouting/doc/drouting.xml | 4 +- modules_k/drouting/doc/drouting_admin.xml | 6 +- modules_k/drouting/doc/drouting_devel.xml | 2 +- 4 files changed, 451 insertions(+), 386 deletions(-) Diff: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commitdiff;h=9d8…

15 years, 6 months

git:master: tm: suspended transaction was not always replied and freed

by Miklos Tirpak

Module: sip-router Branch: master Commit: 090243564b4d54be19991457f7e014c46a131e06 URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=0902435… Author: Miklos Tirpak <miklos(a)iptel.org> Committer: Miklos Tirpak <miklos(a)iptel.org> Date: Thu Jul 2 18:37:56 2009 +0200 tm: suspended transaction was not always replied and freed - When the suspended transaction continued but no new UAC was added to the transaction, for example all the new branches tried to reach a blacklisted destination, then there was no error message returned and the transaction was never freed. --- modules/tm/t_suspend.c | 40 +++++++++++++++++++++++++++++++++++++++- 1 files changed, 39 insertions(+), 1 deletions(-) diff --git a/modules/tm/t_suspend.c b/modules/tm/t_suspend.c index 3c22ab9..2a0cb05 100644 --- a/modules/tm/t_suspend.c +++ b/modules/tm/t_suspend.c @@ -39,6 +39,7 @@ #include "h_table.h" #include "t_lookup.h" #include "t_fwd.h" +#include "t_funcs.h" #include "timer.h" #include "t_suspend.h" @@ -145,6 +146,8 @@ int t_continue(unsigned int hash_index, unsigned int label, Not a huge problem, fr timer will fire, but CANCEL will not be sent. last_received will be set to 408. */ + reset_kr(); + /* fake the request and the environment, like in failure_route */ if (!fake_req(&faked_req, t->uas.request, 0 /* extra flags */, uac)) { LOG(L_ERR, "ERROR: t_continue: fake_req failed\n"); @@ -171,9 +174,44 @@ int t_continue(unsigned int hash_index, unsigned int label, /* update the flags */ t->uas.request->flags = faked_req.flags; + if (t->uas.status < 200) { + /* No final reply has been sent yet. + * Check whether or not there is any pending branch. + */ + for ( branch = 0; + branch < t->nr_of_outgoings; + branch++ + ) { + if ((t->uac[branch].request.buffer != NULL) + && (t->uac[branch].last_received < 200) + ) + break; + } + + if (branch == t->nr_of_outgoings) { + /* There is not any open branch so there is + * no chance that a final response will be received. + * The script has hopefully set the error code. If not, + * let us reply with a default error. + */ + if ((kill_transaction_unsafe(t, + tm_error ? tm_error : E_UNSPEC)) <=0 + ) { + LOG(L_ERR, "ERROR: t_continue: " + "reply generation failed\n"); + /* The transaction must be explicitely released, + no more timer is running */ + UNLOCK_REPLIES(t); + t_release_transaction(t); + t_unref(t->uas.request); + return 0; + } + } + } + UNLOCK_REPLIES(t); - /* release the transaction */ + /* unref the transaction */ t_unref(t->uas.request); return 0;

15 years, 6 months

SF.net SVN: openser:[5893] branches/1.5/packaging/debian/copyright

by Henning Westerholt

Revision: 5893 http://openser.svn.sourceforge.net/openser/?rev=5893&view=rev Author: henningw Date: 2009-07-02 14:27:45 +0000 (Thu, 02 Jul 2009) Log Message: ----------- - extend copyright notice a bit Modified Paths: -------------- branches/1.5/packaging/debian/copyright This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

15 years, 6 months

git:master: core: fix EXTRA_DEBUG information about second via in parse_msg

by Henning Westerholt

Module: sip-router Branch: master Commit: da589b753539ebecfc2294b86bdcfe4bd09d64fb URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=da589b7… Author: Henning Westerholt <henning.westerholt(a)1und1.de> Committer: Henning Westerholt <henning.westerholt(a)1und1.de> Date: Thu Jul 2 12:38:48 2009 +0200 core: fix EXTRA_DEBUG information about second via in parse_msg --- parser/msg_parser.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/parser/msg_parser.c b/parser/msg_parser.c index 5132112..18e17df 100644 --- a/parser/msg_parser.c +++ b/parser/msg_parser.c @@ -625,7 +625,7 @@ int parse_msg(char* buf, unsigned int len, struct sip_msg* msg) #ifdef EXTRA_DEBUG /* dump parsed data */ if (msg->via1){ - DBG(" first via: <%.*s/%.*s/%.*s> <%.*s:%.*s(%d)>", + DBG("first via: <%.*s/%.*s/%.*s> <%.*s:%.*s(%d)>", msg->via1->name.len, ZSW(msg->via1->name.s), msg->via1->version.len, @@ -645,7 +645,7 @@ int parse_msg(char* buf, unsigned int len, struct sip_msg* msg) DBG ("\n"); } if (msg->via2){ - DBG(" first via: <%.*s/%.*s/%.*s> <%.*s:%.*s(%d)>", + DBG("second via: <%.*s/%.*s/%.*s> <%.*s:%.*s(%d)>", msg->via2->name.len, ZSW(msg->via2->name.s), msg->via2->version.len,

15 years, 6 months

SF.net SVN: openser:[5892] branches

by Henning Westerholt

Revision: 5892 http://openser.svn.sourceforge.net/openser/?rev=5892&view=rev Author: henningw Date: 2009-07-02 10:35:20 +0000 (Thu, 02 Jul 2009) Log Message: ----------- - fix EXTRA_DEBUG information about second via in parse_msg Modified Paths: -------------- branches/1.3/parser/msg_parser.c branches/1.4/parser/msg_parser.c branches/1.5/parser/msg_parser.c This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

15 years, 6 months

SF.net SVN: openser:[5889] trunk/etc/kamailio.cfg

by Inaki Baz Castillo

Revision: 5889 http://openser.svn.sourceforge.net/openser/?rev=5889&view=rev Author: ibc_sf Date: 2009-07-01 11:35:07 +0000 (Wed, 01 Jul 2009) Log Message: ----------- - Revert of last commit in kamailio.cfg (it was incorrect). Modified Paths: -------------- trunk/etc/kamailio.cfg This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

15 years, 6 months

git:master: nathelper (k): fix for flag 'l'

by Daniel-Constantin Mierla

Module: sip-router Branch: master Commit: d4796d9a04562e3bc9d45acd68393eb418828e98 URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=d4796d9… Author: Daniel-Constantin Mierla <miconda(a)gmail.com> Committer: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Thu Jul 2 11:44:58 2009 +0200 nathelper (k): fix for flag 'l' - fix SF: #2814137 - handle re-invite with force_rtp_proxy("l"); - patch by Marcus Hunger --- modules_k/nathelper/nathelper.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/modules_k/nathelper/nathelper.c b/modules_k/nathelper/nathelper.c index b9c59a6..01ae520 100644 --- a/modules_k/nathelper/nathelper.c +++ b/modules_k/nathelper/nathelper.c @@ -2608,7 +2608,7 @@ force_rtp_proxy(struct sip_msg* msg, char* str1, char* str2, int offer) if (to_tag.len == 0) return -1; create = 0; - if (swap != 0) { + if (swap != 0 || (msg->first_line.type == SIP_REPLY && offer != 0)) { tmp = from_tag; from_tag = to_tag; to_tag = tmp;

15 years, 6 months

git:master: nathelper(k): fix for very long payload-type

by Daniel-Constantin Mierla

Module: sip-router Branch: master Commit: 8c97cfc35fcaf17f67810f98725bb636601bb38b URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=8c97cfc… Author: Daniel-Constantin Mierla <miconda(a)gmail.com> Committer: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Thu Jul 2 11:47:29 2009 +0200 nathelper(k): fix for very long payload-type - fix SF: #2813924: publishing very long payload-type to the rtp-proxy overflows a buffer - patch by Marcus Hunger --- modules_k/nathelper/nathelper.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/modules_k/nathelper/nathelper.c b/modules_k/nathelper/nathelper.c index 01ae520..8c71ef9 100644 --- a/modules_k/nathelper/nathelper.c +++ b/modules_k/nathelper/nathelper.c @@ -2779,7 +2779,9 @@ force_rtp_proxy(struct sip_msg* msg, char* str1, char* str2, int offer) * a comma-separated list. */ for (cp = payload_types.s; - cp < payload_types.s + payload_types.len; cp++) { + (cp < payload_types.s + payload_types.len) && + (cp1 - (char *)v[1].iov_base < sizeof(opts) - 1); + cp++) { if (isdigit(*cp)) { *cp1 = *cp; cp1++;

15 years, 6 months

git:master: avpops: merge avpops modules

by Klaus Darilion

Module: sip-router Branch: master Commit: 90d2ba0236c7cb4a72ba074fbec7a86a89024e37 URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=90d2ba0… Author: Klaus Darilion <klaus.mailinglists(a)pernau.at> Committer: Klaus Darilion <klaus.mailinglists(a)pernau.at> Date: Thu Jul 2 11:35:22 2009 +0200 avpops: merge avpops modules - delete ser's avpops module (does not compile anyway) - use K's avpops module as common avpops module - Note: probably many of the avpops commands are not needed anymore as AVPs can be used directly in config script --- {modules_k => modules}/avpops/Makefile | 0 {modules_k => modules}/avpops/README | 0 {modules_k => modules}/avpops/avpops.c | 0 {modules_k => modules}/avpops/avpops_db.c | 0 {modules_k => modules}/avpops/avpops_db.h | 0 {modules_k => modules}/avpops/avpops_impl.c | 0 {modules_k => modules}/avpops/avpops_impl.h | 0 {modules_k => modules}/avpops/avpops_parse.c | 0 {modules_k => modules}/avpops/avpops_parse.h | 0 {modules_k => modules}/avpops/doc/Makefile | 0 {modules_k => modules}/avpops/doc/avpops.xml | 0 {modules_k => modules}/avpops/doc/avpops_admin.xml | 0 modules_s/avpops/Makefile | 17 - modules_s/avpops/README | 489 --------- modules_s/avpops/avpops.c | 853 -------------- modules_s/avpops/avpops_db.c | 319 ------ modules_s/avpops/avpops_db.h | 75 -- modules_s/avpops/avpops_impl.c | 1158 -------------------- modules_s/avpops/avpops_impl.h | 132 --- modules_s/avpops/avpops_parse.c | 587 ---------- modules_s/avpops/avpops_parse.h | 56 - modules_s/avpops/doc/Makefile | 4 - modules_s/avpops/doc/avpops.xml | 72 -- modules_s/avpops/doc/faq.xml | 81 -- modules_s/avpops/doc/functions.xml | 371 ------- modules_s/avpops/doc/params.xml | 222 ---- modules_s/avpops/ser-avpops.cfg | 24 - 27 files changed, 0 insertions(+), 4460 deletions(-) Diff: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commitdiff;h=90d…

15 years, 6 months

[ openser-Bugs-2814137 ] force_rtp_proxy bug

by SourceForge.net

Bugs item #2814137, was opened at 2009-06-29 20:29 Message generated for change (Comment added) made by miconda You can respond by visiting: https://sourceforge.net/tracker/?func=detail&atid=743020&aid=2814137&group_… Please note that this message will contain a full copy of the comment thread, including the initial issue submission, for this request, not just the latest update. Category: modules Group: ver 1.5.x >Status: Closed >Resolution: Accepted Priority: 5 Private: No Submitted By: Marcus Hunger (marcushunger) >Assigned to: Daniel-Constantin Mierla (miconda) Summary: force_rtp_proxy bug Initial Comment: force_rtp_proxy seems to handle re-invite wrong, resulting in one-way-audio. ---------------------------------------------------------------------- >Comment By: Daniel-Constantin Mierla (miconda) Date: 2009-07-02 12:33 Message: Patch applied to 1.5. Thanks. ---------------------------------------------------------------------- Comment By: Marcus Hunger (marcushunger) Date: 2009-07-01 12:14 Message: sdp is in invite and 200 ok. the thing is, nearly the same config still worked in 1.2-branch, but after an upgrade to 1.4 it stopped. it happens always, no matter if the callee or caller does the reinvite. the following trace shows what happens. the rtpproxy-port in invite and reply is 41324. they must differ as they are used for different media-streams. 217.10.1.1 - sip-proxy & rtpproxy 172.20.21.2 - sip-proxy doing rtpproxy_* stuff 217.10.66.165 - uac 217.10.1.2 - uas U 217.10.1.1:5060 -> 172.20.21.2:5060 INVITE sip:3993942p0@217.10.1.2 SIP/2.0. Via: SIP/2.0/UDP 217.10.1.1:5060;branch=z9hG4bK78fccbad7757B75A. Via: SIP/2.0/UDP 192.168.234.24;rport=61020;received=217.10.66.165;branch=z9hG4bK78fccbad7757B75A. From: "Erika" <sip:3993942e1@sipgate.de>;tag=D6DBB0FD-C198C6EA. To: <sip:10@sipgate.de;user=phone>;tag=as3804b1ae. Route: <sip:172.20.21.2;lr=on>. CSeq: 4 INVITE. Call-ID: e521bbee-f129e0af-f6e95884(a)192.168.234.24. Contact: <sip:3993942e1@217.10.66.165:61020>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078. Supported: 100rel,replaces. Allow-Events: talk,hold,conference. Max-Forwards: 69. Content-Type: application/sdp. Content-Length: 297. X-hint: rr-enforced. X-nathint: nat. . v=0. o=- 1246290867 1246290869 IN IP4 192.168.234.24. s=Polycom IP Phone. c=IN IP4 192.168.234.24. t=0 0. m=audio 2262 RTP/AVP 9 0 8 18 101. a=sendrecv. a=rtpmap:9 G722/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=direction:active. # U 172.20.21.2:5060 -> 217.10.1.1:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 217.10.1.1:5060;branch=z9hG4bK78fccbad7757B75A. Via: SIP/2.0/UDP 192.168.234.24;rport=61020;received=217.10.66.165;branch=z9hG4bK78fccbad7757B75A. From: "Erika" <sip:3993942e1@sipgate.de>;tag=D6DBB0FD-C198C6EA. To: <sip:10@sipgate.de;user=phone>;tag=as3804b1ae. CSeq: 4 INVITE. Call-ID: e521bbee-f129e0af-f6e95884(a)192.168.234.24. Content-Length: 0. . # U 172.20.21.2:5060 -> 217.10.1.2:5060 INVITE sip:3993942p0@217.10.1.2 SIP/2.0. Via: SIP/2.0/UDP 172.20.21.2;branch=z9hG4bKe2fc.9dcfa89.0. Via: SIP/2.0/UDP 217.10.1.1:5060;branch=z9hG4bK78fccbad7757B75A. Via: SIP/2.0/UDP 192.168.234.24;rport=61020;received=217.10.66.165;branch=z9hG4bK78fccbad7757B75A. From: "Erika" <sip:3993942e1@sipgate.de>;tag=D6DBB0FD-C198C6EA. To: <sip:10@sipgate.de;user=phone>;tag=as3804b1ae. CSeq: 4 INVITE. Call-ID: e521bbee-f129e0af-f6e95884(a)192.168.234.24. Contact: <sip:3993942e1@217.10.66.165:61020>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078. Supported: 100rel,replaces. Allow-Events: talk,hold,conference. Max-Forwards: 68. Content-Type: application/sdp. Content-Length: 315. X-hint: rr-enforced. X-nathint: nat. . v=0. o=- 1246290867 1246290869 IN IP4 192.168.234.24. s=Polycom IP Phone. c=IN IP4 217.10.1.1. t=0 0. m=audio 41324 RTP/AVP 9 0 8 18 101. a=sendrecv. a=rtpmap:9 G722/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=direction:active. a=nortpproxy:yes. # U 217.10.1.2:5060 -> 172.20.21.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.20.21.2;branch=z9hG4bKe2fc.9dcfa89.0;received=172.20.21.2. Via: SIP/2.0/UDP 217.10.1.1:5060;branch=z9hG4bK78fccbad7757B75A. Via: SIP/2.0/UDP 192.168.234.24;rport=61020;received=217.10.66.165;branch=z9hG4bK78fccbad7757B75A. From: "Erika" <sip:3993942e1@sipgate.de>;tag=D6DBB0FD-C198C6EA. To: <sip:10@sipgate.de;user=phone>;tag=as3804b1ae. Call-ID: e521bbee-f129e0af-f6e95884(a)192.168.234.24. CSeq: 4 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces, timer. Contact: <sip:3993942p0@217.10.1.2>. Content-Length: 0. . # U 217.10.1.2:5060 -> 172.20.21.2:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.20.21.2;branch=z9hG4bKe2fc.9dcfa89.0;received=172.20.21.2. Via: SIP/2.0/UDP 217.10.1.1:5060;branch=z9hG4bK78fccbad7757B75A. Via: SIP/2.0/UDP 192.168.234.24;rport=61020;received=217.10.66.165;branch=z9hG4bK78fccbad7757B75A. From: "Erika" <sip:3993942e1@sipgate.de>;tag=D6DBB0FD-C198C6EA. To: <sip:10@sipgate.de;user=phone>;tag=as3804b1ae. Call-ID: e521bbee-f129e0af-f6e95884(a)192.168.234.24. CSeq: 4 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces, timer. Contact: <sip:3993942p0@217.10.1.2>. Content-Type: application/sdp. Content-Length: 329. . v=0. o=root 647471219 647471221 IN IP4 217.10.1.2. s=sipgate VoIP GW. c=IN IP4 217.10.1.2. t=0 0. m=audio 17732 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. ## U 172.20.21.2:5060 -> 217.10.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 217.10.1.1:5060;branch=z9hG4bK78fccbad7757B75A. Via: SIP/2.0/UDP 192.168.234.24;rport=61020;received=217.10.66.165;branch=z9hG4bK78fccbad7757B75A. From: "Erika" <sip:3993942e1@sipgate.de>;tag=D6DBB0FD-C198C6EA. To: <sip:10@sipgate.de;user=phone>;tag=as3804b1ae. Call-ID: e521bbee-f129e0af-f6e95884(a)192.168.234.24. CSeq: 4 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces, timer. Contact: <sip:3993942p0@217.10.1.2>. Content-Type: application/sdp. Content-Length: 347. . v=0. o=root 647471219 647471221 IN IP4 217.10.1.2. s=sipgate VoIP GW. c=IN IP4 217.10.1.1. t=0 0. m=audio 41324 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. a=nortpproxy:yes. ---------------------------------------------------------------------- Comment By: Klaus Darilion (klaus_darilion) Date: 2009-07-01 09:53 Message: do you have a SIP trace? Is it maybe related to late offer (SDP in 200 OK and ACK)? Does the bug happens always or only if the reINVITE is sent by the callee? ---------------------------------------------------------------------- Comment By: Nobody/Anonymous (nobody) Date: 2009-06-30 16:15 Message: in detail, the sdp is rewritten with the wrong port for this direction of the session. it's the same as the answer's which does not work. to build the query for the rtp-proxy from- and to-tag the have to be reversed in this case. ---------------------------------------------------------------------- Comment By: Marcus Hunger (marcushunger) Date: 2009-06-30 12:45 Message: happens when i handle a reinvite with rtpproxy_offer("l") ---------------------------------------------------------------------- Comment By: Daniel-Constantin Mierla (miconda) Date: 2009-06-29 21:19 Message: When is this happening? ---------------------------------------------------------------------- You can respond by visiting: https://sourceforge.net/tracker/?func=detail&atid=743020&aid=2814137&group_…

15 years, 6 months

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