search for: callident

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 find something in the

Ok, I've got the trace translater running above unify at the moment. When I try to access the directory giving me trouble the 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

Hello Anyone have a working copy of Fail2ban asterisk filter asterisk.conf for Asterisk 16 running PJSIP. I have tried 10 different filters but none of them 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

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

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

Hello all, Has anyone have magicjack working with their asterisk? I had patched chan_sip.c with some code that allows asterisk to do the md5 hash that mjmd5 proxy does. * shows that it is registered with magicjack, but incoming calls are not even hitting my * box and outgoing calls get congestion. Here is my relevant configs. I did do a ton of google searching, but it all points to it should

Hello all, I tried the dev list, 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

Hello, I tried to configure a very simple case of Asterisk using SIP userA --- Asterisk server ---- userB sip.conf [userA] type=friend username=userA host=dynamic nat=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,

When calling Originate Action, it rings my phone. If I do not answer, I receive a Channel Event: Hangup, followed by receiving an OriginateResponse event with a Failure Response, Reason 3. My phone continues to ring. If I answer the phone at this point, it attempts to answer, but does not succeed. Looking at the asterisk debug, it appears to destroy the SIP dialog for the call. It also

> Next step is packet capture to see if there is a clue as to the cause of the > failure in the SIP signalling. Right, I see the following when running SIP Debug. Looks to me like the phones are expecting the server to do the conference mixing, which I guess it would do in CallManager? <--- SIP read from TCP:xxx.xxx.xxx.xxx:50604 ---> REFER sip:xxx.xxx.xxx.xxx SIP/2.0 Via:

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

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

Hi all, I am using Digium PCI board to receive PSTN call through 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

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:

I have a working Asterisk 13.1.0 running, and I am trying to configure a SIP trunk for outbound and inbound calling, and a DID for the Asterisk server, which is used for incoming calls from PSTN. I configured my SIP.US trunks (showing one gateway, gw1, here for brevity, have two: gw1 & gw2, which are both configured on my end): [sonnyGW1] type=registration transport=transport-udp

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]

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

09.09.2016 13:45, Joshua Colp ?????: > Dmitry Melekhov wrote: >> 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 >>

Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update-version-9/ On Tue, Jan 20, 2015 at 11:16 AM, Jordan Cook - Gyron Networks < jordan.cook at gyron.net> wrote: > > Next step is packet capture to see if there is a clue as to the cause of > the > > failure in the SIP signalling. > > Right, I

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