search for: 5551212

...ts 1 and 5, and the carrier claims they are all configured the same. The error I'm concerned with it this: -- Forcing restart of channel 0/21 on span 2 since channel reported in use See error in context below. -- Executing Dial("IAX2/unipoint-4@unipoint-4/1", "Zap/G1/5551212") in new stack -- Called G1/5551212 -- Channel 0/23, span 1 got hangup -- Zap/23-1 is busy -- Hungup 'Zap/23-1' == Everyone is busy/congested at this time (1:1/0/0) -- Executing Dial("IAX2/unipoint-4@unipoint-4/1", "Zap/G2/5551212") in...

...conf.. exten => _8.,1,SetVar(CALLFILENAME=${UNIQUEID}--${CALLERIDNUM}--${EXTEN:1}--${TIMESTA MP}) exten => _8.,2,Monitor(wav,${CALLFILENAME},m) exten => _8.,3,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) ; Console log... -- AGI Script Executing Application: (DIAL) Options: (Local/85551212/|30|HS(60605520)) -- Setting call duration limit to 60605520 seconds. -- Executing SetVar("Local/85551212@default-72ef,2", "CALLFILENAME=--spa2002--5551212--20040912-173057") in new stack -- Called 85551212/ -- Executing Monitor("Local/85551212@default-7...

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120 Priority: 1 Asterisk first goes and dials the Channel parameter, SIP/5551212 at provider. This is where it gets confusing. You have no control over what happens here. The actions don't...

Hi! I am searching for the description of the available dialstrin options for the DAHDI channel (and also other channel types). I am not looking for outdated voip-info links, but for the authoritative source, e.g. something like "core show application Dial" Does such thing exists? thanks Klaus

Here is the setup: SIP/3044 -> SetCallerID(5551212) -> Call out PRI The CDR shows a src of 5551212. That is a lie! The src of that call was not 5551212, the source was 3044! The "translated source" of that call was 5551212. How can I get "real" source of this call and not some faky nonsense? The "reason" behind u...

Does anyone know if realtime extensions allows extensions in the format callerid/extension yet? ie the extensions.conf file allows you to do: 5551212/1000 => exten ... and it matches against extension 1000 when the caller id is 5551212. Last time I checked, realtime didn't support this yet. That's a show stopper for us. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipe...

...quot;all circuits are busy now. please try your call again latter" followed by the congestion tone. Instead, I want this to busy ring and then hang up without any message. Here is a snippet from site A: ... [2014-07-09 09:56:16] VERBOSE[21606][C-0000dab7] app_dial.c: -- Called DAHDI/g5/5551212 [2014-07-09 09:56:17] VERBOSE[21606][C-0000dab7] app_dial.c: -- DAHDI/i7/5551212-411b is proceeding passing it to SIP/260-0000a2f1 [2014-07-09 09:56:17] VERBOSE[21606][C-0000dab7] app_dial.c: -- DAHDI/i7/5551212-411b is ringing [2014-07-09 09:56:17] VERBOSE[21606][C-0000dab7] app_dial.c:...

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --

...logic is in the same context on all boxes e. Queue1's dial plan logic is referred to via 3 different DUNDi contexts weighted according to which server is the primary, secondary, and tertiary host server for the user agents (UA1,2, and 3) f. So queue1, assigned the phone number of 5551212, is assigned to the Primary DUNDi context on Astbox1 with the weight of 0 g. Then queue1 is assigned to the secondary DUNDi context on Astbox2 with the weight of 100 and to the tertiary DUNDi context on Astbox3 with the weight of 200 h. So let's say we make a call from an User Age...

...es connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. I see this in the CLI: -- Executing [s at macro-dialout-trunk:19] Dial("SIP/801-09b6e498", "DAHDI/g1/5551212|300|") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/5551212 -- Channel 0/1, span 1 got hangup, cause 28 -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s at macro-dialout-trunk:20] Goto("SIP/801-09b6e498&q...

...in a future release. Please use 'sip set debug' instead. Audio is at 192.168.10.2 port 17492 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.10:5060: INVITE sip:5551212 at 10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport From: "Shawn Ip" <sip:user at 192.168.10.2>;tag=as25dd7670 To: <sip:3538379 at 10.0.0.10> Contact: <sip:user at 192.168.10.2> Call-ID: 0a72cba50ae922e206a7f65c69bc4670 at 192.168.10.2 CS...

...in a future release. Please use 'sip set debug' instead. Audio is at 192.168.10.2 port 17492 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.10:5060: INVITE sip:5551212 at 10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport From: "Shawn Ip" <sip:user at 192.168.10.2>;tag=as25dd7670 To: <sip:3538379 at 10.0.0.10> Contact: <sip:user at 192.168.10.2> Call-ID: 0a72cba50ae922e206a7f65c69bc4670 at 192.168.10.2 CS...

...in a future release. Please use 'sip set debug' instead. Audio is at 192.168.10.2 port 17492 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.10:5060: INVITE sip:5551212 at 10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport From: "Shawn Ip" <sip:user at 192.168.10.2>;tag=as25dd7670 To: <sip:3538379 at 10.0.0.10> Contact: <sip:user at 192.168.10.2> Call-ID: 0a72cba50ae922e206a7f65c69bc4670 at 192.168.10.2 CS...

Anybody ever run into a case where the Sipura Dial Plan will not work with the S0 option to immediately connect? My Dial plan reads (*xx|[3469]11S0|0|00|[2-9]xxxxxxS0|1xxx[2-9]xxxxxxS0) and I can dial ONLY then numbers in the dial plan so I know that it works. For some reason when I dial 5551212 1212121212 It does not dial for a while and then it dials 555 1212 Anyone have any ideas? Thanks Michael Blood -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050927/b1001571/attachment.htm

I'm receiving the following error over and over, adnauseam: Oct 1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce received from 'CNAME-CID <sip:5551212@192.168.1.X>' Does anyone know what "stale nonce" is? Thanks! Paul Conn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051002/5f4162c0/attachment.htm

...uting NoOp("Zap/1-1", "======================> Chan Name jf_linksys") in new stack -- Executing NoOp("Zap/1-1", "======================> Channel Users context longdistance_users") in new stack -- Executing Dial("Zap/1-1", "Local/5551212@longdistance_users/n") in new stack IAXPEER() Seems to be broken or I don't know how to use it properly. -- Executing NoOp("SIP/jf_linksys-08f20548", "======================> Chan Type IAX2") in new stack -- Executing NoOp("SIP/jf_linksys-08f20548&quo...

...o the format of the message coming from the H323 channels included w/ Asterisk we were unable to use our gatekeeper. For a quick solution we tried the OH323 channel drivers and can receive inbound calls from the parent gatekeeper. We are trying to do a dial to gatekeeper... I am trying exten => 5551212,1,Wait,2 exten => 5551212,2,Dial,OH323/5551212 But I am not sure if this is the correct protocol... Please help

Hello, Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/25402ade/attachment.htm

...logic is in the same context on all boxes e. Queue1's dial plan logic is referred to via 3 different DUNDi contexts weighted according to which server is the primary, secondary, and tertiary host server for the user agents (UA1,2, and 3) f. So queue1, assigned the phone number of 5551212, is assigned to the Primary DUNDi context on Astbox1 with the weight of 0 g. Then queue1 is assigned to the secondary DUNDi context on Astbox2 with the weight of 100 and to the tertiary DUNDi context on Astbox3 with the weight of 200 h. So let's say we make a call from an User Age...

Hi All, I have a Cisco 2600 PRI gateway being hosted on an Asterisk server. The PRI on the cisco is pointing to a customer legacy PBX, the SIP VoIP side of the cisco is pointing to an Asterisk server (1.2.X). In Asterisk, the SIP peer is setup with callerid="some name"<5551212> In a SIP call from the cisco to asterisk, there is no CID name info in SIP debug, so Asterisk doesn't seem to inject CID name info for that SIP Channel on pass through calls. Asterisk does have the CID number and account number and user info and all other variable associated with the SIP...