[asterisk-users] ARI originated PjSIP channel changes state to UP while called party is still RINGING
Hello, A call originated from ARI (using ari-py), changes state to UP while the called party is still ringing. The bearer is a PjSIP trunk. I am wondering if this is caused by any kind of early media or incompatibility between my Asterisk and remote SBC but I cannot confirm anything for now since my carrier says the trunk is compliant with VoIP standards. This behaviour is really problematic for my app so I need to resolve it as soon as possible. Do I need to deploy any AMD system? or reconfigure the DIAL command? Any pointer to any possible cause/fix of the problem is very appreciated. Thank you. Best Regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-Kannel integration project version 0.0.1 release notes
Hello, We are proud to announce the first release 0.0.1 version of kannel-asterisk integration project. The goal of this project is to allow asterisk users to use kannel capabilities like SMS sending and receiving. Please visit https://asterisk-kannel.sourceforge.io/ for more information. You can download the release files at: https://sourceforge.net/projects/asterisk-kannel/files/ This version 0.0.1 includes an asterisk app module called app_mt.c which can be used from the dial plan to send SMS MT (mobile terminated). ---technical details: app_mt.c is an asterisk module (a dialplan app called mt) that uses kannel C API to connect to kannel bearerbox as an smsbox and send sms mt messages. It also integrates a thread for receiving ack's and delivery reports (dlr) from bearerbox. ---requirements: -Asterisk source. -Kannel compiled libs and header files (compilation of kannel is not covered here). ---config: -actually the config is done in the source code itself. Adjust the following parameters to fit your setup before compiling and linking the app: static char* dflt_bb_host = "#";//default kannel bearerbox IP address static long dflt_bb_port = 13001;//kannel bearerbox smsbox-port port static int dflt_bb_ssl = 0;//default kannel bearerbox smsbox-port ssl let it 0 if you don't want to use ssl static char* dflt_smsbox_id = "astb";//default smsbox id static char* dflt_service = "csvc";//default service name static char* dflt_account = "supacc";//default account name static char* dflt_from = "18555";//default sender number static char* dflt_to = "1";//default receiver number static char* dflt_smsc_id = "fake-smsc-1";//default smsc-id used to route the sms static char* dflt_dlr_url = "http://127.0.0.1:40001";//default dlr url static int dflt_dlr_mask = 31;//default dlr mask static char* dflt_sms = "Dialplan extension 400 get executed!";//default sms text ---compiling: compilation is similar to any other asterisk module. Just copy the source file to asterisk apps folder, modify your toolchain by adding kannel header files and libs locations. Compile asterisk as usual. you will get app_mt.so generated. ---using: -# cp app_mt.so /usr/lib/asterisk/modules -# asterisk -x "module load app_mt.so" -modify your dialplan to add a test extension for app_mt: exten => 400,1,mt() same => n,Hangup() -call extension 400 from your device, an sms mt will be sent to the receiver number configured above. ---roadmap: -read default config parameters from file. -pass sms parameters from the dialplan. -send sms from cli/manager/rest/...etc. -...etc. Any suggestion is welcome. Any feedback is welcome. Best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail2ban does not work for my asterisk installation
Thank you doctor whom, It is working for me now. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail2ban does not work for my asterisk installation
Thanks for your reply. My configuration is correct. It works with ssh: many attacks have been stopped. Also, the config has worked for asterisk one time: I have seen that in the fail2ban.log file. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fail2ban does not work for my asterisk installation
The failregex statement in my jail.conf file is: * failregex* = NOTICE.* .*: Registration from '.*' failed for 'HOST' - Wrong password NOTICE.* .*: Registration from '.*' failed for 'HOST' - No matching peer found NOTICE.* .*: Registration from '.*' failed for 'HOST' - Username/auth name mismatch NOTICE.* .*: Registration from '.*' failed for 'HOST' - Device does not match ACL NOTICE.* HOST failed to authenticate as '.*'$ NOTICE.* .*: No registration for peer '.*' (from HOST) NOTICE.* .*: Host HOST failed MD5 authentication for '.*' (.*) NOTICE.* .*: Registration from '.*' failed for 'HOST' - ACL error (permit/deny) This is a log entry in /var/log/asterisk/full that shows the scan being performed: *2010-08-01 07:00:13 NOTICE[22540] chan_sip.c: Registration from '123456sip:123...@' failed for '193.158.62.48' - ACL error (permit/deny)* The problem is that fail2ban does not detect this attack that was performed for an amount of time of about half an hour. Please help me identify the problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] My Switch is being attacked using sip scanner tool (Service Abuse Attack)
An attacker is scanning my Asterisk Switch to gain illegitimate access to VoIP call functionality. Using a sip scanning tool, *it* sends REGISTERs with random identities. And when it discovers one identity subscribed in my switch, it tries to authenticate with random passwords using this user name. For the moment, I have replaced this account. And also blocked the IP it has used but each time it tries to use another IP to scan again. Following is a sample REGISTER request sent by it to my switch (I have hidden some info). REGISTER sip:xx.xx.xx.xx SIP/2.0 *Via: SIP/2.0/UDP 127.0.1.1:5061;branch=x**-x**;rport* Content-Length: 0 From: x sip:xx...@xx.xx.xx.xx Accept: application/sdp *User-Agent: friendly-scanner* To: x sip:xx...@xx.xx.xx.xx *Contact: sip:1...@1.1.1.1 sip%3a...@1.1.1.1* CSeq: 1 REGISTER Call-ID: 4244603463 Max-Forwards: 70 Please help me resolve this problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX calling presentation null
Hello all, I am getting a strange behaviour of IAX protocol in an IAX trunk set up for one of our clients. the calling presentation is equal to 0 : *Calling presentation: 0x00* Wireshark presents the call as if the from (caller) is null. It does not seem that there is any config in iax.conf that fixes that. Please help fix that. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] About testing Call transfer in asterisk
Hello, Can you explain how to test blind transfer in asterisk. Here is my test case that hasn't succeeded: I have configured blindxfer = # in features.conf. I have called an iax user from my iax softphone. The called party responds to the call, and tries to transfer the call by clicking the # key followed by the number of another iax extension where I want to transfer the call to. But nothing happened. Please help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Customizing Asterisknow distribution
Hello, I want to modify asterisknow distribution by adding, removing or editing software. How can I do that and recompile a new distribution and put it in a new iso. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voipmonitor.org
Hello, First, thank you for your great job. I want to know why you have choosed to calculate only MOS-LQE. Why you have only used G107. Is that model suitable for VoIP operators to have a calculated QoS value so they can confirm their quality. Thanks again and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voipmonitor.org
Thank you Martin, So the MOS-LQE does not inform bout payload itself but predicts the MOS based on networks metrics and P862 and P863 uses also payload (voice) to calculate the MOS. Is it true what I have understood. Best regards. On Sat, May 8, 2010 at 1:17 PM, Martin Vit v...@lam.cz wrote: Hello, I've choosen only MOS-LQE because it is calculated only on network parameters, which is loss, burstinnes and delay (which is converted to loss by jitterbuffer simulator). It does not takes into account voice (payload). There is no effective objective methods (today) which predicts MOS. Only ITU-T P.862 and P.563 which is patented and it can analyze only 20 seconds samples. I've tried implementing P.563 and it is not usable for real live use, only for automated tests which is not in my interest now (and because of patents). I've calibrated MOS-LQE with polynomial functions using P.862 PESQ. I will write more on voipmonitor.org documentation once I've found more time. I'm using voipmonitor on central gateway and succesfully monitoring all SIP traffic and filtering calls by the worst MOS. So yes, you can use that tool for measuring quality of IP network in realtime. If you save PCAP files, you can analyze it with wireshark in more depth. On Sat, May 8, 2010 at 1:42 PM, mosbah abdelkader mosbah.abdelka...@gmail.com wrote: Hello, First, thank you for your great job. I want to know why you have choosed to calculate only MOS-LQE. Why you have only used G107. Is that model suitable for VoIP operators to have a calculated QoS value so they can confirm their quality. Thanks again and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voipmonitor.org
Ok. Thank you for your help. On Sat, May 8, 2010 at 1:55 PM, Martin Vit v...@lam.cz wrote: On Sat, May 8, 2010 at 2:34 PM, mosbah abdelkader mosbah.abdelka...@gmail.com wrote: Thank you Martin, So the MOS-LQE does not inform bout payload itself but predicts the MOS based on networks metrics yes exactly. LQE is Listen Quality Emodel (E-model is parametric model which takes into account some more parameters. I've used static parameters except for loss and burstiness. So if your network is stable and you want to measure MOS, there is no way how to do that on unknown samples. You can do only automated tests. and P862 and P863 uses also payload (voice) to calculate the MOS. Is it true what I have understood. yes, P.862 (PESQ) compare two samples. Original and degraded (and about 20 seconds). P.563 does not need original sample and can predict only degraded sample (only about 20 seconds). It cannot analyze whole conversation. Both methods is suited for automated tests with specific samples. These objective methods compare new codecs, transmittion path etc. etc.. It will never work as real live passive monitoring. I've used P.862 to calibrate MOS-LQE. MV Best regards. On Sat, May 8, 2010 at 1:17 PM, Martin Vit v...@lam.cz wrote: Hello, I've choosen only MOS-LQE because it is calculated only on network parameters, which is loss, burstinnes and delay (which is converted to loss by jitterbuffer simulator). It does not takes into account voice (payload). There is no effective objective methods (today) which predicts MOS. Only ITU-T P.862 and P.563 which is patented and it can analyze only 20 seconds samples. I've tried implementing P.563 and it is not usable for real live use, only for automated tests which is not in my interest now (and because of patents). I've calibrated MOS-LQE with polynomial functions using P.862 PESQ. I will write more on voipmonitor.org documentation once I've found more time. I'm using voipmonitor on central gateway and succesfully monitoring all SIP traffic and filtering calls by the worst MOS. So yes, you can use that tool for measuring quality of IP network in realtime. If you save PCAP files, you can analyze it with wireshark in more depth. On Sat, May 8, 2010 at 1:42 PM, mosbah abdelkader mosbah.abdelka...@gmail.com wrote: Hello, First, thank you for your great job. I want to know why you have choosed to calculate only MOS-LQE. Why you have only used G107. Is that model suitable for VoIP operators to have a calculated QoS value so they can confirm their quality. Thanks again and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SS7 over an FXO interface
Hello, Is it possible to transfer ss7 signaling over an FXO interface. I need to setup an ss7 test system composed by two Asterisk based IP-PBX systems with anlog interfaces only (FXO and FXS). I want to know if it is possible to connect the two IP-PBX as following: - FXS interface in PBX1 - connected to - FXO interface in PBX2 = used to transport ss7 signaling. - FXS interface in PBX2 - connected to - FXO interface in PBX1 = used to transport voice between the two PBXs. This connection can be replaced by a simple SIP trunk. Is this scenario possible with libss7 and asterisk. If yes, please give some instructions and tips. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Converting an audio file to a .gsm format
Hello all, have anyone an idea about converting an audio file (.wav, .mp3, .au,...) to a .gsm audio file to use it as a voicemail file with Asterisk. Thanks. Abdelkader Mosbah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Locating Asterisk documentation after installation
Hello all, After installing Asterisk, i have installed the docs by make progdocs. But i don't know where to locate this documentation. please Help. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use OpenVPN with Asterisk
Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use OpenVPN with Asterisk
Hello, Is the OpenVPN the ideal solution to set a tunnel between two asterisk servers or there is a better solution. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use OpenVPN with Asterisk
Hello, Have i to install OpenVPN in each Asterisk server or it is enough to install it in one side only?. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re : Connecting two Asterisk servers with a framerelay
Hello, To connect Asterisk to Frame relay network, have i to use the wildcard TE110P. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two Asterisk servers with a frame relay
Hello, As we know, to connect Asterisk to PSTN network, we must use a PCI card containing FXS and FXO modules like Digium TDM400P. Now to connect Asterisk to a Frame Relay network what is the PCI card that we need? Is the Ethernet adapter only is enough? or i have to buy another type of PCI card?. Thanks. Mosbah Abdelkader. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting two Asterisk servers with a frame relay connection
Hello all, I have to connect two Asterisk servers with a frame relay connection but i do not know what is the hardware to use and how to connect them. Have anyone an idea about that. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two Asterisk servers with a frame relay connection
Hello, Have i to buy an asterisk card like TDM400P to connect the two asterisk servers with frame relay. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.18 problem
hello, I have installed asterisk 1.2.18 in suse 10.2. After typing asterisk in the terminal command line (i don't think that asterisk runs when doing this) i type asterisk -r but the response is Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?). how to solve this. thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
