search for: sip_pvt

...3330) at channel.c:671 f = {frametype = 4, subclass = 1, datalen = 0, samples = 0, mallocd = 0, offset = 0, src = 0x0, data = 0x0, delivery = {tv_sec = 0, tv_usec = 0}, prev = 0x0, next = 0x0} #3 0x408fc2d9 in __sip_autodestruct (data=0x81be208) at chan_sip.c:1315 p = (struct sip_pvt *) 0x81be208 #4 0x08056c3e in ast_sched_runq (con=0x8172f28) at sched.c:373 current = (struct sched *) 0x8174868 tv = {tv_sec = 1135275568, tv_usec = 989877} x = 0 res = 1083432672 #5 0x40927e28 in do_monitor (data=0x0) at chan_sip.c:11253 res = 0 s...

...roblem we have here is to find out the best option (even one that results on some internal Asterisk files changing) that allow us to propagate the SIP header "Call-ID" to both modules described above. The best shot we have until now is to use the "callid" field from the "sip_pvt" structure of SIP channel, what will lead us to two considerable code changes: 1- Propagate the channel to method "realtime_var_get" of our proprietary ARA driver; and 2- Duplication of necessary structs to a header (".h") file so the modules can "navigate" on...

...subclass = 1, datalen = 0, samples = 0, > > mallocd = 0, offset = 0, src = 0x0, data = 0x0, delivery = {tv_sec = > > 0, > > tv_usec = 0}, prev = 0x0, next = 0x0} > > #3 0x408fc2d9 in __sip_autodestruct (data=0x81be208) at chan_sip.c:1315 > > p = (struct sip_pvt *) 0x81be208 > > #4 0x08056c3e in ast_sched_runq (con=0x8172f28) at sched.c:373 > > current = (struct sched *) 0x8174868 > > tv = {tv_sec = 1135275568, tv_usec = 989877} > > x = 0 > > res = 1083432672 > > #5 0x40927e28 in do_moni...

...at code, but why is the function register_verify defined in 2 different files? chan_iax2.c line 3860 static int register_verify(int callno, struct sockaddr_in *sin, struct iax_ies *ies) chan_sip.c line 4869 /*--- register_verify: Verify registration of user */ static int register_verify(struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req, char *uri, int ignore) Regards Greg

Hello, How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough variables in (within) my custom Asterisk application? I can't use chan_sip.c internal structures (such as sip_pvt) in my custom application, because there's no chan_sip.h and I can't include it into my application (maybe there's other way?). I can do like this: exten => _X,1,Set(PEERIP=${SIPCHANINFO(peerip)}) exten => _X,2,custom_app and read PEERIP with pbx_builtin_getvar_he...

...o strongly discourage setting nat per-peer/user when at all possible. * Fixed SendMessage stripping extension from To: header in SIP MESSAGE When using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure. For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc3 Thank you for your c...

...o strongly discourage setting nat per-peer/user when at all possible. * Fixed SendMessage stripping extension from To: header in SIP MESSAGE When using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure. For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc3 Thank you for your c...

...ciejewsk) * --- Fix local channel chains optimizing themselves out of a call. (Closes issue ASTERISK-16711. Reported by Alec Davis) * --- Update a peer's LastMsgsSent when the peer is notified of waiting messages (Closes issue ASTERISK-17866. Reported by Steve Davies) * --- Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt. (Closes issue ASTERISK-19425. Reported by David Cunningham) * --- Send more accurate identification information in dialog-info SIP NOTIFYs. (Closes issue ASTERISK-16735. Reported by Maciej Krajewski) For a full list o...

What does this mean? Asterisk 1.2.13 talking to 1.4.0. (response from 1.4.0.) Yuan Liu

...tion ${SIPCHANINFO(from)} But how to get the "To" field ? I have tried to add some code line into the chan_sip.c ... It works partially ... meaning that, I can add this "to" in SIPCHANINFO funciton, but the result is null. Here is what I have added in chan_sip.c : in structure sip_pvt ( "to" field same as "from" ) in sipchaninfo_function added "to" Line same as "from" function_sipchaninfo_read added "to" line same as "from" So I believe that I have enabled somehow Asterisk to read the value to from the channel ... But...

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

...XXX,3,Wait(1) exten => _9XXXX,4,Dial(IAX2/myserv:mypw@remote-regist-server/${EXTEN:1}) exten => _9XXXX,5,Congestion exten => _9XXXX,105,Playback(tt-monkeysintro) exten => _9XXXX,106,Hangup my chan_sip.c: static int sip_write(struct ast_channel *ast, struct ast_frame *frame) { struct sip_pvt *p = ast->pvt->pvt; int res = 0; if (frame->frametype == AST_FRAME_VOICE) { if (!(frame->subclass & ast->nativeformats)) { --> --> ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %d (read/write = %d/%d)\n", frame->subcla...

...to dig in and see what those lines actually do. Did soneone just get wacky with cut and paste from the peer while loop? Or am I breaking something else. Jon --- chan_sip.c.broken Fri Nov 7 02:17:47 2003 +++ chan_sip.c Fri Nov 7 02:16:23 2003 @@ -3928,8 +3928,8 @@ static int check_user(struct sip_pvt *p, p->callgroup = user->callgroup; p->pickupgroup = user->pickupgroup; p->restrictcid = user->restrictcid; - p->capability = user->capability; -...

...umber does not change from its value in the previous invite. However, the Polycom phone tracks its transactions this way - the branch numbers must be different for new invites. So here's the change: In chan_sip.c, in transmit_reinvite_with_sdp(): static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp, struct ast_rtp *vrtp) { struct sip_request req; if (p->canreinvite == REINVITE_UPDATE) reqprep(&req, p, "UPDATE", 0); else { // BEGIN POLYCOM CHANGE p->branch++; snprintf(p->via, sizeof(p->via), "SIP/2.0/UDP %s:%d;br...

...einstall chan_sip.so, stop and restart asterisk, and you should > be good to go. > -Brian > > --- chan_sip.c.fcs 2003-12-13 14:54:37.000000000 -0800 > +++ chan_sip.c 2005-03-10 11:48:40.000000000 -0800 > @@ -4444,10 +4446,10 @@ > } > > static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader, char *msg, int init) { > - char digest[256]; > + char digest[1024]; > p->authtries++; > memset(digest,0,sizeof(digest)); > - if (reply_digest(p,req, "Proxy-Authenticate", msg, digest, sizeof(digest) )) { &g...

...respid = 0 res = 0 debug = 0 e = 0x7fffff892e79 "sip:10.3.8.1" error = 0 __PRETTY_FUNCTION__ = "handle_incoming" #12 0x0000000809067e4f in handle_request_do (req=0x7fffff892a40, sin=0x7fffff892a30) at chan_sip.c:19466 p = (struct sip_pvt *) 0x802817028 recount = 0 nounlock = 0 lockretry = 10 __PRETTY_FUNCTION__ = "handle_request_do" #13 0x0000000809067876 in sipsock_read (id=0x802811184, fd=12, events=1, ignore=0x0) at chan_sip.c:19372 req = {rlPart1 = 0x7fffff892e70 "REGISTER&...

...d 10020). Now we need to make three small changes to the file /usr/src/asterisk/channels/chan_sip.c 3. First find the function build_contact( ) and insert your ?outside? ip address in the right position, as is indicated below (the original line is commented out): static void build_contact(struct sip_pvt *p) { /* Construct Contact: header */ if (ourport != 5060) snprintf(p->our_contact, sizeof(p->our_contact), "<sip:%s@%s:%d>", p->exten, inet_ntoa(p->ourip), ourport); else // snprintf(p->our_contact, sizeof(p->ou...

...ou need to make a tiny change to the asterisk code to pass the mailbox user in the SIP NOTIFY packet. --- channels/chan_sip.c.orig Thu Jul 14 12:03:18 2005 +++ channels/chan_sip.c Thu Jul 14 12:05:26 2005 @@ -9710,6 +9710,7 @@ /* Called with peerl lock, but releases it */ struct sip_pvt *p; int newmsgs, oldmsgs; + char *s; /* Check for messages */ ast_app_messagecount(peer->mailbox, &newmsgs, &oldmsgs); @@ -9735,6 +9736,10 @@ /* Recalculate our side, and recalculate Call ID */ if (ast_sip_ouraddrfor(&p->sa.sin_addr,&...

I did some searching in the archive, but found only one message with this same question and no answer. Hopefully it's a simple config problem. When the Caller-ID is delivered, it is surrounded by double-quotes, like this: "ATA-57 1" On long caller-id strings, the last character is cut off to make room for the leading double-quote: "BudgeTone 1234 instead of BudgeTone

...ct sip_pkt *) 0x0 owner = <value optimized out> __PRETTY_FUNCTION__ = "dialog_unlink_all" #7 0x00002aaaceac2f5a in dialog_needdestroy (dialogobj=<value optimized out>, arg=<value optimized out>, flags=<value optimized out>) at chan_sip.c:19564 dialog = (struct sip_pvt *) 0x2aab18b892d8 __PRETTY_FUNCTION__ = "dialog_needdestroy" #8 0x000000000044736e in internal_ao2_callback (c=0x1346c4c8, flags=6, cb_fn=0x2aaaceac2d70, arg=0x0, data=0x0, type=DEFAULT, tag=0x0, file=0x0, line=0, func=0x0) at astobj2.c:1102 match = -827576976 __list_head = (struct b...