search for: 18005551212

...-- AGI Script call_log.agi completed, returning 0 -- Hungup 'IAX2/u32218094-1' == Manager 'sendcron' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 -- Executing AGI("Local/18005551212@default-5973,2", "call_log.agi|18005551212") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi +++++ CALL LOG START : |1139506499.6|Local/18005551212@default-5973,2|18005551212|Local|V0210013458000000002|2006-02-10 1:34:59 -- AGI Script call_log.agi...

...I know that much, I just don't know how to fix it. SIP debugging shows that it attempts to retransmit packets to my phone and since it can't, it drops it after 30 seconds. Log snippet: -- Executing [s at macro-dialout-trunk:19] Dial("SIP/203-b7a2b558", "SIP/bw_outbound/+18005551212|300|") in new stack Audio is at <public IP> port 11968 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 216.82.224.202:5060: INVITE sip:+18881231234 at 216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP &...

...ery touchy. Can anyone shed some light on this before I open a ticket? {this is a packet capture of the typical flow of examples #1 and #2, which work correctly} [root@app1 asterisk]# tethereal port 5060 Capturing on eth0 0.000000 128.151.224.35 -> 128.151.224.11 SIP/SDP Request: INVITE sip:18005551212@128.151.224.11;user=phone, with session description 0.000439 128.151.224.11 -> 128.151.224.35 SIP Status: 100 Trying 0.001085 128.151.224.11 -> 128.151.224.35 SIP Status: 180 Ringing 1.980925 128.151.224.11 -> 128.151.224.35 SIP/SDP Status: 200 OK, with session description 2.071965...

...MONITORDIR}/${CALLFILENAME}-rev.gsm ${MONITORDIR}/${CALLFILENAME}.gsm reverse) exten => s,9,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-rev.gsm) exten => s,10,System(sox ${MONITORDIR}/${CALLFILENAME}.gsm -g ${MONITORDIR}/${CALLFILENAME}.wav exten => s,11,NoOp == Spawn extension (sip, 18005551212, 2) exited non-zero on 'SIP/one-8e46' -- Executing Macro("SIP/one-8e46", "record-cleanup") in new stack -- Executing GotoIf("SIP/one-8e46", "0?11:2") in new stack -- Goto (macro-record-cleanup,s,2) -- Executing SetVar("SIP/one-8e4...

Hello everybody, Can someone explain [gGrR] in Dial() function? To dial external extension 18005551212 over channel 2 we will use: Dial(DAHDI/2/18005551212) To dial external extension 18005551212 over one of channel from group of channels (nr 2) we will use: Dial(DAHDI/g2/18005551212) So lets assume that group 2 consists of 5 channels. How does Dial() function choose channel: - randomly? - first f...

...portions of zapata.conf: [trunkgroups] [channels] language=en context=from-pstn pridialplan=unknown signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master ;signaling=em_w switchtype=4ess group=0 channel => 1-23 debug info: -- Executing Dial("SIP/202-2d92", "zap/g0/18005551212") in new stack -- Making new call for cr 32771 -- Requested transfer capability: 0x00 - SPEECH > Protocol Discriminator: Q.931 (8) len=48 > Call Ref: len= 2 (reference 3/0x3) (Originator) > Message type: SETUP (5) > [04 02 80 90] > Bearer Capability (len= 4) [ Ext: 1...

Using CVS-HEAD libpri, CVS-HEAD zaptel, CVS-STABLE asterisk. Everything compiled fine. No problems loading chan_zap.so. Incomming calls to PRI work fine. Outbound is a different story: -- Executing Dial("SIP/64.72.107.4-4122fb40", "ZAP/R1d/18005551212|60") in new stack -- Called R1d/18005551212 -- Channel 0/23, span 1 got hangup Mar 13 13:19:29 WARNING[28835]: chan_zap.c:7149 zt_pri_error: PRI: Call Reference Length not supported: 0 -- Zap/23-1 is circuit-busy -- Hungup 'Zap/23-1' == Everyone is busy/congested at t...

...recent versions of Asterisk either with chan_sip or pj_sip? > > PJSIP does not currently have functionality to allow such a thing. I > believe in chan_sip there have been no changes to remove it. > My DP looks like this: Exten => aaa,1,Dial(SIP/USERNAME:PASSWORD at sip1.myproxy.net/18005551212) and from the logs I get: oice1*CLI> console dial aaa at from-external -- Executing [aaa at from-external:1] Dial("Console/default", "SIP/ USERNAME:PASSWORD at sip1.myproxy.net/18005551212") in new stack [2019-07-09 08:40:54] NOTICE[27159][C-00019e64]: chan_sip.c:30586...

I assume that from your previous post that you are using iconnect Is your register line in the format: Register => 18005551212:1234@213.137.73.178/18005551212 I've had good luck using the IP address vs. the fully qualified hostname. Remember that the register line goes in the [general] section of sip.conf. Also, are you using the latest CVS release of *? -----Original Message----- From: Senad Jordanovic [mailto:sen...

I'm trying to call a conference service, wait 10 seconds, then send the passcode. I've tried ww: Dial(SIP/18005551212ww12345#@sip.com,60,r) The sip channel didn't like that. Added 'p' , still no help. I tried D: Dial(SIP/18005551212 at sip.com,60,rD(12345#) The dtmf is sent too soon. I tried inserting 'ww' but that was just sent. I tried G: exten => 234.1.Dial(SIP/18005551212 at sip.co...

...wo problems: 1) if I dial 800 numbers, like (800)555-1212, I get a bunch of silence and the following in my log: -- Starting simple switch on 'Zap/97-1' -- Executing NoOp("Zap/97-1", "") in new stack -- Executing Goto("Zap/97-1", "intern-post|818005551212|1") in new stack -- Goto (intern-post,817005556226,1) -- Executing Dial("Zap/97-1", "IAX2/myusername:mypassword@iaxtel.com/18005551212@iaxtel") in new stack -- Called myusername:mypassword@iaxtel.com/18005551212@iaxtel -- Call accepted by 69.73.19.178 (forma...

...It contains example data that needs editing to fit your scenario. so the pbxmonitor.db might have (separated by tabs): SIP/2000 Receptionist SIP/2001 Username 2 SIP/2002 Username 3 an internal call might say: Username 2 talking to Receptionist an outgoing call might say: Username 3 talking to 18005551212 an incoming call (already answered) might say: 18005551212 talking to Receptionist It's pretty self explanatory I guess. Run it and hope it does stuff. so, pbxmonitor, in our application, is called from watch, like so: watch -t -n 1 pbxmonitor but you could implement it into a refreshing...

Has anyone made, or have any simple PHP, or HTML interfaces where by a user could enter their number and the number they want to call, and have asterisk bridge the calls?

Hi, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center /n&Local/3 at dial_call_center/n&Local/4 at dial_call_center /n&Local/5 at...

I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216

...t undestand how to deal with extensions.conf? I replaced Dial (ZAP/ ...) with Dial (DAHDI/ ... ) All my inbound calls from DAHDI work the same as ZAP. The outbound calls aren't working. I get: -- Executing [8005551212 at internal:1] Macro("SIP/107-b4e703f0", "dialout-dahdi,18005551212") in new stack -- Executing [s at macro-dialout-dahdi:1] Set("SIP/107-b4e703f0", "CALLERID(number)=781-736-1994") in new stack -- Executing [s at macro-dialout-dahdi:2] Set("SIP/107-b4e703f0", "CALLERID(name)=Jim Duda") in new stack -- Execut...

...f Of tim.mcqueen@qualisys.biz Sent: 11 September 2003 20:14 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 I assume that from your previous post that you are using iconnect Is your register line in the format: Register => 18005551212:1234@213.137.73.178/18005551212 I've had good luck using the IP address vs. the fully qualified hostname. Remember that the register line goes in the [general] section of sip.conf. Also, are you using the latest CVS release of *? -----Original Message----- From: Senad Jordanovic [mailto:sen...

Hi, Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you should be able to dial with SIP credentials in the DP. Is this still possible in recent versions of Asterisk either with chan_sip or pj_sip? TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

I saw that if I add a c to my Dial string as follows: exten => s,3,Dial(Zap/g2c/18005551212) That it will not consider that call as answered until the called party presses #. When the number dialed picks up does a bridge of the call immediately instead of waiting for the # key. I am using a PRI, does that make a difference???? -------------- next part -------------- An H...

I've been searching the archives for the proper "Wait for Dial tone" command in the "Dial(Zap/g1/18005551212)" dial sting. Does anyone have an example of it's use? Arick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040906/6f8b21a6/attachment.htm