Re: [asterisk-users] Emulate and script emulation of users calling in/receiving calls, transferring calls etc
Thanks, looks like: sipp.sourceforge.net supports scripting... Are there sample University Curricula for teaching VOIP with Asterisk or FreeSwitch? On Sun, Oct 16, 2011 at 5:26 AM, Daniel Tryba dan...@tryba.nl wrote: On Sat, Oct 15, 2011 at 08:12:33PM +1100, Alec Taylor wrote: If asterisk or freeswitch would be taught in a classroom environment, is there someway to emulate and script emulation of users calling in/receiving calls, transferring calls etc? The Asterisk Manager Interface (AMI) and callfiles to/from Echo(), voicemail or IVRs! But much easier is to have multiple (soft)phones available to 1 student. -- Daniel Tryba -- _ -- 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 -- _ -- 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] Asterisk 1.8.7 and client outside network
Hi Tarek Le 15/10/2011 20:28, Tarek Sawah a écrit : Hello Daniel First question, do you have a firewall application or hardware installed on the network? The Asterisk server is also the firewall/router, iptables running on it. Second do you have some software similar to fail2ban? Yes, but I put the domain IP in ignoreip list. I checked fail2ban iptables rules, no trace of this IP Third check your IPTABLES if you can post the output of iptables-save would be good. if you can replace the localnet=Asterisk server external IP/32 with externip=Asterisk server external IP/32 I didn't send this info but externalip is setted to Asterisk server external IP/32 then we will be able to check your problem? This setup is working on tens of customers servers (1.2, 1.4 and 1.6), but this is the first one running 1.8 version. The same phone connect perfectly to our 1.6 server in the same conditions, so it's seems something related to 1.8 version. What I don't understand is that (violating IP ) should display the IP but in my case it's blank (or empty). Should domain contain as well the port despite the fact that we have insecure=port,invite? Thanks for your help Daniel Date: Sat, 15 Oct 2011 19:08:10 +0200 From: ad...@tootai.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.7 and client outside network Hi, no clue on this? I found a thread in march from Faisal Hanif having the same problem but no one of the proposed ideas where working (reverse permit/deny, tried with only permit=0.0.0.0/0.0.0.0, aso), no luck :-) I don't now if it's solved for him. If someone had a solution on this, would be great to share ;-) Regards -- Daniel Le 07/10/2011 15:01, Administrator TOOTAI a écrit : Hi, my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and GrandStream) connected from the lan I now want to connect a snom320 from outside but it failed, having always [Oct 7 14:48:04] ERROR[3870]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:13597 parse_register_contact: Domain 'XX.XXX.XXX.XX:2048' disallowed by contact ACL (violating IP ) [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:14306 register_verify: Registration denied because of contact ACL doesn't matter if I connect through a VPN or to the public IP using STUN. My sip.conf: localnet=172.24.0.0/12 localnet=169.254.0.0/255.255.0.0 ; Zero conf local network localnet=Asterisk server external IP/32 autodomain=yes ;allowexternaldomains=yes domain=172.24.30.250 ;Asterisk Server IP domain=Public Hostname domain=Another Public Hostname [309](snom320,ulaw-phone,callgroup1) type=friend insecure=port,invite secret=VoIP2auDIo contactdeny=0.0.0.0/0.0.0.0 contactpermit=XX.XXX.XXX.XX/32 ; External IP from phone, same as disallowed by contact ACL deny=0.0.0.0/0.0.0.0 permit=XX.XXX.XXX.XX/32 nat=yes Any clue? Why violating IP is empty? -- _ -- 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] Asterisk 1.8.7 and client outside network
One more thing can you post your peer's configs as you have it in the config file? and can you register with the same user from within the lan? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Sun, 16 Oct 2011 12:33:27 +0200 From: ad...@tootai.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.7 and client outside network Hi Tarek Le 15/10/2011 20:28, Tarek Sawah a écrit : Hello Daniel First question, do you have a firewall application or hardware installed on the network? The Asterisk server is also the firewall/router, iptables running on it. Second do you have some software similar to fail2ban? Yes, but I put the domain IP in ignoreip list. I checked fail2ban iptables rules, no trace of this IP Third check your IPTABLES if you can post the output of iptables-save would be good. if you can replace the localnet=Asterisk server external IP/32 with externip=Asterisk server external IP/32 I didn't send this info but externalip is setted to Asterisk server external IP/32 then we will be able to check your problem? This setup is working on tens of customers servers (1.2, 1.4 and 1.6), but this is the first one running 1.8 version. The same phone connect perfectly to our 1.6 server in the same conditions, so it's seems something related to 1.8 version. What I don't understand is that (violating IP ) should display the IP but in my case it's blank (or empty). Should domain contain as well the port despite the fact that we have insecure=port,invite? Thanks for your help Daniel Date: Sat, 15 Oct 2011 19:08:10 +0200 From: ad...@tootai.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.7 and client outside network Hi, no clue on this? I found a thread in march from Faisal Hanif having the same problem but no one of the proposed ideas where working (reverse permit/deny, tried with only permit=0.0.0.0/0.0.0.0, aso), no luck :-) I don't now if it's solved for him. If someone had a solution on this, would be great to share ;-) Regards -- Daniel Le 07/10/2011 15:01, Administrator TOOTAI a écrit : Hi, my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and GrandStream) connected from the lan I now want to connect a snom320 from outside but it failed, having always [Oct 7 14:48:04] ERROR[3870]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:13597 parse_register_contact: Domain 'XX.XXX.XXX.XX:2048' disallowed by contact ACL (violating IP ) [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:14306 register_verify: Registration denied because of contact ACL doesn't matter if I connect through a VPN or to the public IP using STUN. My sip.conf: localnet=172.24.0.0/12 localnet=169.254.0.0/255.255.0.0 ; Zero conf local network localnet=Asterisk server external IP/32 autodomain=yes ;allowexternaldomains=yes domain=172.24.30.250 ;Asterisk Server IP domain=Public Hostname domain=Another Public Hostname [309](snom320,ulaw-phone,callgroup1) type=friend insecure=port,invite secret=VoIP2auDIo contactdeny=0.0.0.0/0.0.0.0 contactpermit=XX.XXX.XXX.XX/32 ; External IP from phone, same as disallowed by contact ACL deny=0.0.0.0/0.0.0.0 permit=XX.XXX.XXX.XX/32 nat=yes Any clue? Why violating IP is empty? -- _ -- 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 -- _ -- 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] **OT** Fwd: oFono 1.0 has been released
When oFono launched, I announced the project to other projects that it may compliment. oFono has hit the 1.0 Milestone and has some serious backing if you missed my post a year or so ago and never heard of it. Check it out... Thanks, Steve Totaro -- Forwarded message -- From: Marcel Holtmann mar...@holtmann.org Date: Sun, Oct 16, 2011 at 2:25 PM Subject: oFono 1.0 has been released To: of...@ofono.org Hello everybody, I am pleased to announce that we have released oFono 1.0 this week. This marks a major step for oFono and we consider it fully feature complete for 2G and 3G telephony. oFono is released under GPL version 2 and is 100% open source. It includes support for the majority of data modem vendors and also full voice telephony support for Infineon (now IMC), ST-Ericsson, Nokia/ISI and also Calypso/Freerunner. All standard features including voice calls, supplementary services, text messaging, USSD, SIM Toolkit, network registration, multiple GPRS contexts and many more have been integrated with easy to use D-Bus APIs. With our 1.0 out of the door, we will continue to improve our CDMA support and also integrate LTE into oFono. So stay tuned for new features. The tarballs are not yet available due to the security breach on kernel.org, but will be uploaded as soon as service has been restored. Regards Marcel ___ ofono mailing list of...@ofono.org http://lists.ofono.org/listinfo/ofono -- _ -- 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] PRI E1 call termination issue
Hi List, I have configured TE121PF card in E1 mode. I am using asterisk 1.6 and freepbx 2.7. My card staus turn in to green and looks like sync with the service provider. My service provider is BSNL - India. I have one toll free number for incoming and one land line number for out going calls. Problem : If i am calling to the toll free number, i am getting the ring but that call is not reaching to my asterisk box. Both incoming and outgoing are failure. Please reffer the following informations for understand the issue further. 1. [root@localhost src]# cat /etc/dahdi/system.conf span=1,0,0,CCS,HDB3,CRC4 bchan=1-15,17-31 dchan=16 loadzone=us defaultzone=us 2. cat /etc/asterisk/chan_dahdi.conf ; Copied from DAHDI Module of FreePBX [general] #include chan_dahdi_general.conf [channels] ; include dahdi groups defined by DAHDI module of FreePBX #include chan_dahdi_groups.conf ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf 3. cat /etc/asterisk/chan_dahdi_groups.conf ; [span_1] signalling=pri_cpe switchtype=euroisdn pridialplan=national prilocaldialplan=national group=1 context=from-pstn channel=1-15,17-31 ** *Some CLI outputs* localhost*CLI *dahdi show channel 1 *Channel: 1 File Descriptor: 14 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: Calling TON: 0 Caller ID name: Mailbox: none Destroy: 0 InAlarm: 1 Signalling Type: ISDN PRI Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 1 taps (unless TDM bridged) currently OFF Wait for dialtone: 0ms PRI Flags: PRI Logical Span: Implicit Hookstate (FXS only): Onhook localhost*CLI 2. localhost*CLI *pri show spans* PRI span 1/0: Provisioned, Down, Active localhost*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Overlap Dial: 0 Logical Channel Mapping: 0 Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T202: 1 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000 T313: 4000 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 Overlap Recv: No 3. localhost*CLI *pri set debug on span 1* Enabled debugging on span 1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) Changing from state 5(Awaiting establishment) to 4(TEI assigned) TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) TEI=0 Sending SABME Changing from state 4(TEI assigned) to 5(Awaiting establishment) TEI=0 Sending SABME TEI=0 Sending SABME TEI=0 Sending SABME TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) Changing from state 5(Awaiting establishment) to 4(TEI assigned) TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) TEI=0 Sending SABME Changing from state 4(TEI assigned) to 5(Awaiting establishment) TEI=0 Sending SABME TEI=0 Sending SABME TEI=0 Sending SABME TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) Changing from state 5(Awaiting establishment) to 4(TEI assigned) TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) TEI=0 Sending SABME Changing from state 4(TEI assigned) to 5(Awaiting establishment) TEI=0 Sending SABME TEI=0 Sending SABME TEI=0 Sending SABME TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) Changing from state 5(Awaiting establishment) to 4(TEI assigned) TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) Please some one help me to identify the issue Michael.k -- _ -- 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] Voicemail in asterisk 1.8.7, stop working
Hi All;
I was having asterisk 1.8.5 and I installed 1.8.7, I copied the
extensions.conf, sip.conf, voicemail.conf that I was using them in 1.8.5 to the
new asterisk 1.8.7.
Every thing fine, but now the voicemail is not working at all !! Even does not
display on the consol any thing that it declares to run the Voicemail function,
instead of running the voicemail, it rings some rings and then run to the next
step !! I am using the below configurations:
[macro-voicemail]
exten = s,1,Dial(${ARG1},20)
exten = s-NOANSWER,1,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = s-NOANSWER,2,Goto(IncomingPSTN,t,3)
exten = s-BUSY,1,Voicemail(${MACRO_EXTEN}@Internal,b)
exten = s-BUSY,2,Goto(IncomingPSTN,t,3)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${MACRO_EXTEN})
What is the difference in using asterisk 1.8.5 and 1.8.7 in voicemail part? How
I can troubleshoot?!
Regards
Bilal
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