search for: callid

Hi, To combine two sources of CDR's I want Asterisk to save the SIP callid for all calls. I know there's a variable that contains the SIP CallID value, but is this the callid value of the incoming INVITE message or the outgoing message? Are they the same? (I've not yet checked a trace, I'm sorry for that). I've tried to read chan_sip, but couldn't...

...following gets logged (note, I accessed one which worked first - /home/lozzar, and then mine /home/will): 2008-09-15 20:16:53 C [dict.c:1125:data_to_str] dict: @data=(nil) 2008-09-15 20:16:53 C [dict.c:1125:data_to_str] dict: @data=(nil) 2008-09-15 20:16:53 T [trace.c:1117:trace_lookup] trace: callid: 2 (*this=0x50cd30, loc=0x526768 {path=/, inode=0x5266a0} ) 2008-09-15 20:16:53 W [client-protocol.c:280:client_protocol_xfer] brick1: attempting to pipeline request type(1) op(34) with handshake 2008-09-15 20:16:53 W [client-protocol.c:280:client_protocol_xfer] brick2: attempting to pipeline req...

...show any matches when testing with fail2ban-regex I see date template hits but no matches.... My log [2019-06-06 15:37:20] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" <sip:2405 at asterisk>' failed for '71.127.239.22:65476' (callid: 50670137772977-30593645157868 at 192.168.1.8) - Failed to authenticate [2019-06-06 15:37:52] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '"as100" <sip:as100 at 95.179.170.109>' failed for '188.214.128.172:5076' (callid: 03e7f9d2dcdf425...

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I have 1 cdr record with only inbound call leg callid. I can see all the call legs with ras...

Sirs, According to RFC 2705 (MGCP), these are the parameters that are used in the transactions: ReturnCode, Connection-parameters <-- DeleteConnection(CallId, EndpointId, ConnectionId, [Encapsulated NotificationRequest,] [Encapsulated EndpointConfiguration]) ReturnCode, <-- DeleteConnection( CallId,...

...ash, a2); snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, usednonce, a2_hash); ast_md5_hash(resp_hash, resp); + + + /* To a Magicjack domain */ + if (strstr(uri,"talk4free.com")) + { + char callid[256]; + char newnonce[256]; + char *c; + int i; + ast_copy_string(callid, p->callid, sizeof(callid)); + ast_copy_string(newnonce, p->nonce, sizeof(newnonce)); + + strcat(newnonce, "_"); +...

...st, but got no answer at all. I'm facing some problems with call-id generation in a heavily loaded Asterisk Server. Asterisk is generating same call-id and from tag for different calls (and this is not desirable). Looking at the source code I noticed that rand() is used four times to get a callid. Is that safe enough? Maybe my system lacks of a good random number generator. Is that possible? What is necessary for a linux box (Debian, in my case) to achieve good random numbers (and consequently "good" callids)? Best Regards, Chuck. __________________________________...

...=no context=test [userB] type=friend username=userB host=dynamic nat=no context=test In extensions.conf [test] exten => 1000,1,Dial(SIP/userA) exten => 2000,1,Dial(SIP/userB) I make a call from userA to userB, it works, but I have 2 questions: 1/ By verifing with Wireshark, I see that the CallID of the INVITE message sent from userA to Asterisk is different from the CallID of the INVITE message sent from Asterisk to UserB. Is it possible to configure Asterisk as an "normal SIP proxy" (it just forwards SIP messages)? 2/ After sending the INVITE message to UserB, Asterisk send an...

...s not succeed. Looking at the asterisk debug, it appears to destroy the SIP dialog for the call. It also destroys the RTP instance. When I answer, I receive messages... [Oct 1 15:35:34] DEBUG[3129]: chan_sip.c:6206 find_call: That's odd... Got a response on a call we dont know about. Callid 2c47c6e4740289d90a0d1337261fd704 at 192.168.9.241 [Oct 1 15:35:34] DEBUG[3129]: chan_sip.c:21256 handle_request_do: Invalid SIP message - rejected , no callid, len 715 [Oct 1 15:35:35] DEBUG[3129]: chan_sip.c:6206 find_call: That's odd... Got a response on a call we dont know about. Callid...

...uest+xml Content-Disposition: session;handling=required Content-Id: <9a2a9191 at xxx.xxx.xxx.xxx> <?xml version="1.0" encoding="UTF-8"?> <x-cisco-remotecc-request> <softkeyeventmsg> <softkeyevent>Conference</softkeyevent> <dialogid> <callid>203a07fc-eb4b001c-1bf7ad61-614d38c1 at xxx.xxx.xxx.xxx</callid> <localtag>203a07fceb4b00ed3e4e2321-d9cb1581</localtag> <remotetag>as4a087ee2</remotetag> </dialogid> <linenumber>0</linenumber> <participantnum>0</participantnum> <consu...

Hi, Surely, I must be overlooking something. If I run the following SQL queries I don't get the same number of rows. Is this coherent? mysql> select * from queue_log where queuename = '4010' and FROM_UNIXTIME(time) between 20080308000000 and 20080313145900 group by callid; 357 rows in set (0.01 sec) mysql> select * from cdr where dst = 4010 and calldate between 20080308000000 and 20080313145900 group by uniqueid; 219 rows in set (0.19 sec) Thanks! ____________________________________________________________________________________ Looking for last mi...

No Jeremy, excuse me, the error was in my email. The correct command is /bin/echo "Channel: Local/$1@chiamamezzi-dialout";\ /bin/echo "Variable: callid=123456|number=$1|url=pippo|menuid=FOP|redirectnum=0554202880";\ /bin/echo "Context: chiamamezzi-Wave";\ /bin/echo "Exten: s";\ /bin/echo "Priority: 1";\ /bin/echo "Callerid: Asterisk Automatic Wardial";\ /bin/echo "Timeout: 10000";\ /bin/echo &...

...regular phone line. It is no problem for me to receive calls, but I am not able to obtain the Caller ID if the calls are from the phone line. exten => s,1,Answer() exten => s, n, Verbose(1|incoming number is ${CHANNEL} calling to ${EXTEN} routing to ${phonenum} ) exten => s,n, Verbose(1|callid is ${CALLID(num)}) exten => s,n,Verbose(1|callpres is ${CALLINGPRES}) exten => s,n,Dial(SIP/${phonenum}@voipuser,60) -- Executing [s at incoming:3] Verbose("Zap/1-1", "1|incoming number is Zap/1-1 calling to s routing to ") in new stack -- Executing [s at incom...

I am not getting any caller Id with my standard T-1. Is a standard "T" capable of sending callerid? I don't want to spend time troubleshooting my PBX if Asterisk can't send it down that type of trunk. Jordan -------------- next part -------------- An HTML attachment was scrubbed... URL:

...the request, but is not able to do anything, owing to the following error: [Mar 13 20:12:26] NOTICE[9783]: res_pjsip/pjsip_distributor.c:255 log_unidentified_request: Request from '"SONNY THE MAN " < sip:6175551212 at 67.231.1.110>' failed for '65.254.44.194:5060' (callid: 335817341_133371542 at 67.231.1.110) - No matching endpoint found [Mar 13 20:12:26] NOTICE[9783]: res_pjsip/pjsip_distributor.c:255 log_unidentified_request: Request from '"SONNY THE MAN " < sip:6175551212 at 67.231.1.110>' failed for '65.254.44.194:5060' (callid: 3...

Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"3563" <sip:3563 at 192.168.32.254>' failed for '192.168.32.116:5060' (callid: 0_1409534529 at 192.168.32.116) - No matching endpoint found or [Sep 9 12:56:14] NOTICE[10163] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '"3567" <sip:3567 at 192.168.32.254>' failed for '192.168.32.108:5060' (callid: 0_2410349837 at 192.168...

Hi, I am making calls from the dial plan using the dial application. Due to technical requirements I need to find out the sip call-id used in the dialog initiated by the dial application. I dont see any straight forward way of doing this so I am looking for answers. There is a sip callid session variable but the problems is that dial is a blocking call and the dialog ends when dial returns. I saw a similar post on the users list but there was no apparent solution suggested. Our setup permits simultaneous calls as well so I need to retreive call id's for each call made. I woul...

...ded 13.10 to 13.11.1 today and now I see messages in log: >> >> >> [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request >> 'REGISTER' from '"3563" <sip:3563 at 192.168.32.254>' failed for >> '192.168.32.116:5060' (callid: 0_1409534529 at 192.168.32.116) - No >> matching endpoint found >> >> >> or >> >> [Sep 9 12:56:14] NOTICE[10163] res_pjsip/pjsip_distributor.c: Request >> 'INVITE' from '"3567" <sip:3567 at 192.168.32.254>' failed for >&g...

...session;handling=required > Content-Id: <9a2a9191 at xxx.xxx.xxx.xxx> > > <?xml version="1.0" encoding="UTF-8"?> > <x-cisco-remotecc-request> <softkeyeventmsg> > <softkeyevent>Conference</softkeyevent> <dialogid> > <callid>203a07fc-eb4b001c-1bf7ad61-614d38c1 at xxx.xxx.xxx.xxx</callid> > <localtag>203a07fceb4b00ed3e4e2321-d9cb1581</localtag> > <remotetag>as4a087ee2</remotetag> </dialogid> <linenumber>0</linenumber> > <participantnum>0</participantnu...

Calling linphone from asterisk 13.9.1.: Dial(SIP/<user>@sip.linphone.org) And it works. But on the linphone side the caller is: <extno>@ipaddress or 2502 at 45.123.987.4 Is there any way to make it more descriptive, at least for the sip user name ? I tried setting SIPCALLID, which had no effect. Set(SIPCALLID=Office) Thanks, sean