search for: sip_request

...TERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I disable ulaw the outgoing call doesn't work. disallow=all ;allow=ulaw allow=alaw debug message: File chan_sip.c, Line 5590 (sip_request): Asked to get a channel of unsupported format ULAW while capability is ALAW Why Asterisk doesn't use the SAME codec with outgoing & incoming calls ? In my AS5300, dtmf is configured as dtmf-relay rtp-nte perhaps I should try with h245-signal or h245-alphanumeric ? ALL ideas will be rea...

...erify 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

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

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

...rrors this morning: Dec 8 10:44:07 WARNING[50315282]: rtp.c:829 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:2352 sip_alloc: Unable to create RTP session: Too many open files Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:8024 sip_request: Unable to build sip pvt data for 'xxxxxxxxxx@sip0' Dec 8 10:44:07 NOTICE[50315282]: app_dial.c:743 dial_exec: Unable to create channel of type 'SIP' I don't think it's related to the unreachable peer thing from last year, this machine only has one peer and a TE405p acting...

Is anyone out there using app_dbodbc (http://www.bkw.org/~brian/app_dbodbc.c)? Any problems with it? I was able to get it all working, but it causes * to segfault every now and then. It does not appear to be related to any specific function (ODBCget,ODBCput,ODBCdel,ODBCdelltree). It is 100% repeatable. If I noload the module, everything works fine, but when its running, after calls to any of the

...ver, 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;branch=z9hG4bK%08x", inet_ntoa(p->ourip), ourport, p->branch...

...elay the ringing from the call through the registrar. Anybody know if I'm out of luck? I already looked into writing in support for adding the ability for asterisk to send 302 redirects to certain hosts when they are dialed via Dial(SIP/1231231234@pstn) and couldn't find where the initial sip_request was by the time dial executed sip_call so that I could create the proper 302 response... my C-fu is not so strong... :-)

...o, 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) )) { > + if (reply_digest(p...

...ion limit to 3000 seconds. Sep 2 15:58:12 WARNING[10334]: rtp.c:852 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Sep 2 15:58:12 WARNING[10334]: chan_sip.c:2313 sip_alloc: Unable to create RTP session: Too many open files Sep 2 15:58:12 WARNING[10334]: chan_sip.c:8202 sip_request: Unable to build sip pvt data for '5000131@192.168.0.11:5060' Sep 2 15:58:12 NOTICE[10334]: app_dial.c:1084 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time -- Executing Hangup("SIP/192.168.0.11-436cd6d8", "") in...

Hello everybody! [Introduction] This is a quite long message, but I think the problem is interesting. [The problem] Does anyone know how can I tell Asterisk that a certain user has a certain telephone number (or address)? For example, I have some registered users, but nor the client (X-lite) nor the server (Asterisk) specifies what telephone number has the user. I don't want to

...gt;> Brian >> >> Patch I used: >> >> --- 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",...