search for: 5065

Hello I want to unlist the attached element getting only the first element in each element of the list. The last element of the list looks as this: [[5065]] [[5065]]$Pluv3Meses [1] 274.4 [[5065]]$PluvMesesMedio [1] 378.2667 [[5065]]$Pluv2UltimosMeses [1] 23.33333 So I would like to get for each element of the list the element called Pluv3Meses. The whole list has 5065 elements but when I try to unlist it I am getting 5081 elements I don't know w...

...[10652] logger.c: -- Requested transfer capability: 0x00 - SPEECH Oct 20 13:19:45 DEBUG[8159] channel.c: Avoiding initial deadlock for 'Zap/25-1' Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Called g2/5155 Oct 20 13:19:45 VERBOSE[10652] logger.c: Transmitting (no NAT) to 172.16.8.200:5065: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425957956;received=172.16.8.200 From: "From Desk"<sip:5100@172.16.200.5>;tag=2425948795 To: <sip:5155@172.16.200.5>;tag=as279eb184 Call-ID: 2425954456-c4756-5065@172.16.8.200 CSeq: 2 INVITE User-Agent...

...er I can extract them from an INVITE. The code is: exten => _.,n,Set(custom-id=${SIP_HEADER(custom-id)}) ; exten => _.,n,Set(custom-valid=${SIP_HEADER(custom-valid)}) ; Examples of the INVITE (works) and REFER (doesn't) messages are below. U 147.202.001.001:5060 -> 127.0.0.1:5065 INVITE sip:0116499123123 at 127.0.0.1:5065 SIP/2.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK8b04.6e642c74.0 To: <sip:0116499123123 at 127.0.0.1:5065> From: <sip:6495566778 at domain.co.nz>;tag=119438778730084 CSeq: 1 INVITE Call-ID: 119438778730084 Content-Length: 142 User...

...16:9004 192.168.23.3:9004 ---->>192.168.17.17:9004 But i am getting below error on my syslog ################################################################################ Jul 21 20:25:38 testlvs nanny[5064]: starting LVS client monitor for 192.168.23.3:9004 Jul 21 20:25:38 testlvs nanny[5065]: starting LVS client monitor for 192.168.23.3:9004 Jul 21 20:25:43 testlvs pulse[5040]: gratuitous lvs arps finished Jul 21 20:25:47 testlvs nanny[5065]: READ from 192.168.17.16:9004 was too short Jul 21 20:25:47 testlvs nanny[5064]: READ from 192.168.17.17:9004 was too short Jul 21 20:25:53 testl...

...have done my best effort but still not resolved i am adding a callerid in script still same please help me if some one can IP1:5060 -> IP2:5060 INVITE sip:0423347871787 at IP2:5060 SIP/2.0..Via: SIP/2.0/UDP IP1:5060;branch=z9hG4bK- 966123148--16781 75694--693700493-4-..Via: SIP/2.0/UDP IP1:5065;branch=z9hG4bK00749b6d;rport..Call-Id: 4f8a33b207cd65f060b083b57 804dac8 at IP1..To <804dac8 at 117.20.20.234..To>: <sip:3347871787 at IP1>.. * *From: "asterisk"<sip:asterisk at IP1:5065>;tag=as0cae0b** see the last part this is what that i want to change here in from...

Hi folks, To provide MWI, * will send out a sip:notify message to the UA. The originator of this message is asterisk, as shown below; NOTIFY sip:1001@www.mysipproxy.com:5065 SIP/2.0 Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21 From: "asterisk" <sip:asterisk@66.121.xxx.yyy>;tag=as0ffb1bdc <=============== To: <sip:1001@www.mysipproxy.com:5065> Contact: <sip:asterisk@66.121.xxx.yyy> Call-ID: 6fa70c151dce64a339367cc2612c9f87@6...

Hi, Does anyone has configured Asterisk to connect to BabyTel (a SIP Provider in Canada) ? Here is my sip.conf (I'm behind a firewall and I already opened port 5060 and 5065 (udp and tcp) to my Asterisk server): [general] port = 5065 context = Test insecure = very register => 1514XXXXXXX:password@sip.babytel.ca When starting Asterisk, the sip registration failed after 5 connecting attempt to sip.babytel.ca. Any clue ? Thanks -- WayComm Wayne Veilleux ing., GC...

...ds to. But in CDR, I can only find the from(1011) and destination(PSTN number). I can't find UA 9999 from this CDR so I can't charge to UA 9999. It seems unreasonable. I use asterisk -rvvvvvvvv and "sip debug" to debug my sip channel. And I build my sip proxy(5060) and asterisk(5065) on the same machine. I make a call from 1011 to 9999 on sip proxy, sip proxy forwards this call to "0939749001". And this 0939749001 will be sent to asterisk by sip proxy. sip ua(1011) => sipproxy => sip ua 9999 ( call forward 0939749001) ||...

...There is no modification rule at my CISCO. (It will not change any dialed number) UA1 ==> SER ==> UA2 (SIP to SIP) UA1 ==> SER ==> Asterisk ==> CISCO 5300 ==> PSTN (SIP to PSTN) port:5060 port:5065 port:5060 IP:xxx.xxx.190.248 IP:xxx.xxx.190.243 (On the same server) (On another server) I know how to forward a call from ser to CISCO 5300. And I have done it ever. UA1 ==> SER ==> CISCO 5300 ==> PSTN Now I modify the ser.cfg, and want to fo...

...ying to put the same values in asterisk, but there seems to be one field that doesn't seem to exist in asterisk - that of outbound proxy all suggestions welcome SIP headings account = user password = secret registrar = host = registrar.provider.com outbound proxy = ?? = nat.provider.com:5065 If I put in an extra field of port=5065 it doesn't register (in sip show registry) with either of the above addresses in the host box. Putting registrar.provider.com in the register => string and nat.provider.com in the host makes outbound calls fail, putting registrar.provider.com in the...

Hi all, I am running an Asterisk server behind a NAT. I want to forward the calls from PSTN to a SIP phone (no nat and also an asterisk). I set the externip and localnet in sip.conf already. I opened the ports in my firewall. (I changed SIP port from 5060 to 5065 and limited the rtp port to 12000-13000) However, I just can't call out. I've always received SIP/2.0 404 Not Found. My sip.conf looks somewhat like this [general] context=default ; Default context for incoming calls bindport=5065 ; UDP Port to bind to (SIP standar...

...mail : gary.holzer@acedivision.com.au mobile : 0417 094 921 direct : (08) 8273 5903 switch : (08) 8271 5455 fax : (08) 8271 1055 w: www.acedivision.com.au & www.cmebookings.com postal: PO Box 17 ~ Fullarton ~ SA 5063 location: Glenside Campus ~ 226 Fullarton Rd ~ Glenside ~ SA 5065 ACEDGP - "Providing Health Care Intelligence"

Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed to make it work, all my terminals spa Cisco 5XX look my cli [Jul 8 11:09:16] ERROR[14733]: pjsip:0 <?>: tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:16] WARNING[14733]: pjsip:0 <?>: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=48024 (tdta0x7f53c000dcb0)!

...246. The following is the information contained in the vocal tag. [vocal] type=friend ; either "friend" (peer+user), "peer" or "user" callerid=Test 1 <12346> host=10.117.4.236 ; we have a static but private IP address port=5065 This indicates that the calls will be received from the Vocal server running on host 10.117.4.236 on port 5065 (port where Marshall server is running). The following details are included in the extensions.conf file so that calls originating from vocal can be answered by the extension 12346....

I'm trying to configure SIP trunking. Now, I'm referencing "Asterisk the definitive guide", 4th ed. While I don't have the page handy, I was reading the suggestion to try SIP to SIP before proceeding to outside connectivity. I'm aware that SIP trunking is a construct, but am, obviously, learning the system. What I'd like to do is from the CLI "ping"

I trying to receive a SIP call and have ring a analog phone. I have a TDM11B card with FXS(green) module in line 1. I have Sip server "SER" setup to accept a SIP call, add a 970 extension to uri and set to asterisk SIP server on port 5065. When I send a SIP call from "kphone a soft SIP phone" running to sip://wally.world@cci.net "SER" picks call ok and changes uri to sip://9703653113@cci.net before relaying it to asterisk. Asterisk does pickup the SIP call OK errors with Everyone is busy/congested. Specfical...

...that often these situations didn't exist when users were connected directly to our VoIP provider, voiptalk.org. They have something which they call a NAT-Traversal Gateway (see item 6 at http://www.voiptalk.org/products/voiptalkfaq.html), which one configures as the outgoing proxy, using port 5065. Does anyone have any idea what this NAT-Traversal Gateway could be? Naturally, I'm asking this in the hope that I can install something similar on our server to solve our NAT issues. Thanks in advance for your help. Best regards, David Shirley

Hi All, I have CentOS 4.2 with ser 0.9.6 and asterisk 1.2.4. Ser is listening on 5060 and asterisk on 5065. The setup is that people use serweb to create an account and register a phone. Their calls are routed from ser to asterisk and then inbound on IAX2. The server has a public and an internal interface. The real FQDN of the server is nmibwksip3.nexusmgmt.com and it has cnames of pbx and nexphone. T...

Hi all I fancied playing with SER and * on the same box. So i thought i'd just change the default sip port for * in sip.conf [general] port = 5065 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) restarted * and now when i issue a > ]# netstat -anp |grep 5060 > udp 0 0 0.0.0.0:5060 0.0.0.0:* 9453/asterisk Its still...

Hi All, sorry if this was discussed before but I didn''t find any solution for my problem. Test site.pp consists of 1 line: file { "/tmp/bar": audit => content } exec { ''/usr/bin/true'': refreshonly => true, subscribe => File[''/tmp/bar''] } and it produces this error every time: err: /Stage[main]//Node[localhost]/File[/tmp/bar]: