similar to: Unable to capture CallerID through Zap

Greetings, I'm trying to continue to do some processing after a TIMEOUT (absolute). In my dialplan below, when a call comes in to [default], I call macro-phonenum and pass it a timeout of 20 seconds. macro- phonenum sets TIMEOUT(absolute), then loops saying the phone number that was called (in MACRO_EXTEN). When the timeout expires I want to call my macro-hangup (so it can say

Hi, I have a Linksys SPA 3102 using as ATA. The call routing is : Phone -> PSTN -> SPA 3102 -> SIP Proxy -> Asterisk The problem I am having is that when the phone hangs up, SPA 3102 can't detect it and relay the CANCEL message. Is this problem with my SPA 3102 config or it just works like that by default? Thanks in advance for your help. Regards, Mark -------------- next

Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. Best Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081119/e74ef6b1/attachment.htm

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

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

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 capture the destination number through the ZAP channel exten => s, n, Verbose(1|destination to ${EXTEN} ) ${EXTEN} returns 's' instead of the actual destination number. Since I have multiple phone numbers, I want to be able to route

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

Hi I am working on deploying voip for my company and would like to seek some advice on the number of E1 lines we need to rent. Our telco told us that there can be at most 30 concurrent channels on an E1 line. Typically, what is the maximum number of DIDs that we can allocate to that E1 line before users get frequent "all lines are busy"? We are running a support center with mostly

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,

Hi, I'm having a problem calling international numbers with debian's Asterisk 1.0.7 w/ zaptel 1.0.9 in Moscow. Russia doesn't seem to have touchtone dialing, so pulsedial is enabled on my TDM400P interface. Local numbers work fine, but when it comes to long distance or international, I'm lost. The prefix for these should be 8 (wait for dialtone) 10 (country code) (city code)

> 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:

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 We want optimize my extensions file conf on asterisk 11.4.0 : We have a big quantity of extensions, all are same "design": ; Destination: Gambia Type: Fixe exten => _00220X.,1,Set(CDR(CodeCom)=BUS-GMB) exten => _00220X.,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) exten => _00220X.,3,Set(CALLERID(all)=${NUMID}) exten =>

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 simulating the sending of fax using sendfax through voip to reach an Asteria server via ZAP/1 ( PSTN phone line ) which then route call to a fax machine at ZAP/2. It seems like I am not able to establish any handshake with the physical fax machine using the sendfax program. Does anyone know why that happens and how to fix it? The scenario will be deployed in remote location in the

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 >>

Dear friends, Asterisk's voicemail functions work fine for me, but I am having difficulty storing the voice messages inside mysql. My real-time CDR recording works so I assume the odbc connection is fine. The voicemail.conf I have is : [general] format = wav attach = yes dbuser=root dbpass=sqlpass dbhost=localhost dbname=asterisk odbcstorage=asterisk odbctable=voicemessages Asterisk shows

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

Dear friends, I am having problem with running a sample php and I can't figure out why. I can run the sample.php using CLI but when I run it inside the dialplan it does not work. Can someone please suggest the config problem that I may have made? dommy:/var/lib/asterisk/agi-bin# php sample.php #!/usr/bin/php5 -q VERBOSE "Here we go!" 2 VERBOSE "Call from - Calling

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