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  • How to send REGISTER request periodically from my SIP client to Asterisk server using Asterisk Manag

    - by Prashant
    Hi, I am using Asterisk 1.4 server and I have created a desktop client using the Asterisk.NET Library. I am able to log into the AMI (as a manager) using Asterisk.Net, but I cannot find a way to send the REGISTER command using the AMI, to share my client's location information with the server. I want to know an AMI or a CLI command that can send a REGISTER request to the Asterisk Server. Thanks

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  • Asterisk SIP digest authentication username mismatch

    - by Matt
    I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue of older 3com phones that come without a SIP image. The 3com phones are communicating SIP with the Asterisk, but are unable to register because they present a digest username value that doesn't match what Asterisk thinks it should. As an example, here are the relevant lines from a successful registration from a soft phone: Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1cac3853" Phone responds: Authorization: Digest username="2321", realm="asterisk", nonce="1cac3853", uri="sip:192.168.254.12", algorithm=md5, response="d32df9ec719817282460e7c2625b6120" For the 3com phone, those same lines look like this (and fails): Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Phone responds: Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" Basically, Asterisk wants to see a username in the Digest username field of 2321, but the 3com phone is sending sip:[email protected] Anyone know how to tell asterisk to accept this format of username in the digest authentication? Here is the sip.conf info for that extension: [2321] deny=0.0.0.0/0.0.0.0 disallow=all type=friend secret=1234 qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes [email protected] host=dynamic dtmfmode=rfc2833 dial=SIP/2321 context=from-internal canreinvite=no callerid=device <2321 allow=ulaw, alaw call-limit=50 ... and for those interested in the grit, here is the debug output of the registration attempt: REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (no NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) confbridge*CLI REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 403 Authentication user name does not match account name Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) Thanks for your input!

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  • Asterisk SIP digest authentication username mismatch

    - by Matt
    I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue of older 3com phones that come without a SIP image. The 3com phones are communicating SIP with the Asterisk, but are unable to register because they present a digest username value that doesn't match what Asterisk thinks it should. As an example, here are the relevant lines from a successful registration from a soft phone: Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1cac3853" Phone responds: Authorization: Digest username="2321", realm="asterisk", nonce="1cac3853", uri="sip:192.168.254.12", algorithm=md5, response="d32df9ec719817282460e7c2625b6120" For the 3com phone, those same lines look like this (and fails): Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Phone responds: Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" Basically, Asterisk wants to see a username in the Digest username field of 2321, but the 3com phone is sending sip:[email protected] Anyone know how to tell asterisk to accept this format of username in the digest authentication? Here is the sip.conf info for that extension: [2321] deny=0.0.0.0/0.0.0.0 disallow=all type=friend secret=1234 qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes [email protected] host=dynamic dtmfmode=rfc2833 dial=SIP/2321 context=from-internal canreinvite=no callerid=device <2321 allow=ulaw, alaw call-limit=50 ... and for those interested in the grit, here is the debug output of the registration attempt: REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (no NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) confbridge*CLI REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 403 Authentication user name does not match account name Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) Thanks for your input!

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  • asterisk queues priority and penalty

    - by MealstroM
    queues.conf shared_lascall=yes strategy=rrmemory wrapuptime=15 A1,A2,A3 are members of 2 queues: queue1(Q1) and queue2(Q2) A3 has penalty 3 in Q1 where min/max penalty are 0/3 and A3 has penalty 0 at Q2 where min/max penalty are 0/3. A3 has just ended a call and is on wrapuptime pause. User1 (U1) enters Q1 with priority 10, and user2 (U2) enters Q2 with priority 15. A3 wrapuptime ends. What user U1 or U2 will be served first?

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  • Asterisk not playing custom sounds on Ubuntu Server 11.04

    - by jochy2525
    I've installed Asterisk on my Ubuntu Server, all works fine, excepts playing the custom sounds. Asterisk sounds work, but this file I've uploaded does not play (on other servers it works, it is a .WAV PCM 16bit 8000). Here is some log output: [Feb 6 22:55:45] WARNING[11045] file.c: File custom/sohoitsoluciones does not exist in any format [Feb 6 22:55:45] WARNING[11045] file.c: Unable to open custom/sohoitsoluciones (format 0x4 (ulaw)): No such file or directory [Feb 6 22:55:45] WARNING[11045] app_playback.c: ast_streamfile failed on SIP/Out4903-0000001d for custom/sohoitsoluciones How can I get Asterisk to play a custom sound?

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  • Asterisk terminating outbound call when picked up, sends 'BYE' message

    - by vo
    I'm running Asterisk 1.6.1.10 / FreePBX 2.5.2.2 and I've got an outbound trunk setup. Everything use to work fine until recently (perhaps due to upgrade to FC12 or other things I'm not sure). Anyway the setup does not appear to have issues registering and setting up the call, RTP packets go both ways and you can hear the ringing from the other side. However it appears that when the call is picked up or thereabouts, the incoming RTP packets cease. Upon closer inspection with Wireshark, there are these particular packets that seem to be the cause: trunk->asterisk SIP/SD Status: 200 OK, with session description asterisk->trunk SIP Request: ACK sip:<phone>@trunk:6889 asterisk->trunk SIP Request: BYE sip:<phone>@trunk:6889 [..about a dozzen RTP packets in/outbound..] trunk->asterisk SIP Status: 200 OK, CSeq: 104 Bye [..outbound RTP continues, phone is silent..] Then the inbound RTP packets cease, however the asterisk logs dont show any activity at this point. The last entry reads 'SIP/ is answered SIP/'. Then when you hangup the extension, you get asterisk->trunk SIP Request: BYE sip:<phone>@trunk:6889 trunk->asterisk SIP Status: 481 Call Leg/Transaction does not exist My trunk peer settings in FreePBX are: username=<user> fromuser=<user> canreinvite=no type=friend secret=<pass> qualify=no [qualify yes produces 401/forbidden messages] nat=yes insecure=very host=<sip trunk gateway> fromdomain=<sip trunk gateway> disallow=all context=from-pstn allow=ulaw dtmfmode=inband Under sip_general_custom.conf i have stunaddr=stun.xten.com externrefresh=120 localnet=192.168.1.1/255.255.255.0 nat=yes Whats causing Asterisk to prematurely end the call and still think the call is in progress? I have no idea where to look next.

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  • Skype Connect as SIP/Trunk for Asterisk

    - by Kaurin
    First off: I'm not sure if this should be on superuser or here. I have recently built a few Asterisk boxes with OpenVOX FXO/FXS ports little or no trouble. My current project is building an Asterisk box with SIP trunks. My current employer insisted on getting Skype Business/Skype connect for that purpose. After reviewing the Skype Connect plan, I agreed, because I thought it is going to be straightforward: Purchase G729 licences and setup SIP trunk/trunks. Boy was i wrong :) Here is the setup: The setup is for calling US numbers only via skype (we got skype US minute bundles in skype connect) AsteriskNOW - Asterisk 1.4 + asterisk-gui Trunks: SIP Trunk configured with Skype Connect - shows as registered Users: 2 test extensions. Both work fine when calling each other, voicemail etc works fine too The asterisk box is behind a Mikrotik router which i configured to forward all relevant ports: 5060-5090 UDP, 10000-20000 UDP. When trying out an extension outside of my LAN, it worked. I could make calls to the other extension. Outgoing rule: _NXXXXXXXXX Strip:0 Prepend:+1 Use skype trunk Inbound rule: Trunk: Skype Pattern: s Destination: Extension1 (6210) Here is the output of asterisk CLI (-rvvvvv) with outgoing calls: http://pastebin.com/eWVpL72e you can see the circuit-busy response when using trunk1 (skype) When calling my Skype Connect number from the outside, I get nothing in the logs. Can anyone with Skype Connect / Asterisk experience help out? :)

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  • cli commands not working asterisk on ubuntu

    - by Mian Khurram Ijaz
    hi guys today my first day on asterisk on ubuntu. I installed vmware and then ubuntu and then on ubuntu i am running asterisk. i started the asterisk server successfully by following a tutorial. After making few changes into the sip.conf i want to reload the sip.conf i issue the command sip reload and nothing happens neither the commands to restart the asterisk server work actually the commands do not exists can some please throw light or point to right direction. thanks

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  • Asterisk failing at startup after upgrading to asterisk18

    - by Supratik
    I was using asterisk16 and asterisk16-skypeforasterisk, which was working fine. I have recently upgraded to asterisk18 and asterisk18-skypeforasterisk, after that I am receiving the following error message. Asterisk ended with exit status 1 Asterisk died with code 1. Asterisk could not start! Use 'tail /var/log/asterisk/full' to find out why. When I checked the log I got the following messages. codec_g729a.c: == Found total of 11 G.729 licenses translate.c: empty buf size, you need to supply one Now, if I remove the /var/lib/asterisk/licenses folder it works fine. Can you please tell me what could be the issue here ? Warm Regards Supratik

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  • Can we manage incoming call number landing on asterisk?

    - by user194469
    I am using Asterisk 1.4.2 in two different machine. I have configured some extensions in asterisk. When any caller, dial my extension number with local number then if I see asterisk console (asterisk -r) then incoming number is starting with 0, but if caller dial same extension number using STD number then in asterisk console (asterisk -r), incoming number is staring with 0091 (here 91 is country code). Could I change this setting or is there any standard for asterisk for local, STD or ISD number?

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  • Asterisk - Trying to use call files to create a conference call between two dynamic numbers

    - by Hank
    I'm trying to setup an Asterisk system that will allow me to create a conference call between two dynamic numbers. It seems I can use 'call files' to make Asterisk initiate the call without needing an incoming call - http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out This example seems to be what I'd need: Channel: SIP/mytrunk/12345678 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: callme Extension: 800 Priority: 2 I can generate this file with some scripting language and then place it into the Asterisk Call File folder. The problem I'm having is: How do I call out to two numbers and join them in a conference call? The MeetMe plugin/extension seems to be what I need in terms of conference calling, I'm just unsure as to how I'd use the two together and join them. Also, is it possible to have multiple 2-person conference calls at the same time? Is setting this up as simple as setting aside X amount of 'channels' in the meetme.conf?

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  • Asterisk - Trying to use call files to create a conference call between two dynamic numbers

    - by Hank
    I'm trying to setup an Asterisk system that will allow me to create a conference call between two dynamic numbers. It seems I can use 'call files' to make Asterisk initiate the call without needing an incoming call - http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out This example seems to be what I'd need: Channel: SIP/mytrunk/12345678 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: callme Extension: 800 Priority: 2 I can generate this file with some scripting language and then place it into the Asterisk Call File folder. The problem I'm having is: How do I call out to two numbers and join them in a conference call? The MeetMe plugin/extension seems to be what I need in terms of conference calling, I'm just unsure as to how I'd use the two together and join them. Also, is it possible to have multiple 2-person conference calls at the same time? Is setting this up as simple as setting aside X amount of 'channels' in the meetme.conf?

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  • Asterisk/FreePBX: Allow other Google Talk clients to ring when using motif module

    - by larsks
    I've recently installed FreePBX to act as a link between a SIP soft phone and my Google Talk account. It was easy to set up and outbound calls work just fine, but I've run into two problems with inbound calls that I'm not sure how to resolve. I'm using an inbound route to forward all calls from Google to my soft phone. If the soft phone is not currently registered, Asterisk answers and immediately generates a fast-busy signal (reporting CHANUNAVAIL in the logs), and the call is lost. If the soft phone is registered, Asterisk "answers" the call before rining the soft phone, which means that other Google Talk clients never ring (since from their perspective someone has answered the call). For solving (1) seems like I could use the ChanIsAvail() function (or this answer) to prevent Asterisk from answering in the event that the phone isn't registered. However, I'm not sure what to do about (2), because the behavior I want is for Asterisk to not "answer" the call until I answer the call on the soft phone. How do I configure Asterisk (ideally within the FreePBX framework) such that I can continue to receive calls at other Google Talk clients in addition to forwarding them to a SIP phone?

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  • Suggestions for a SOHO/Home Asterisk system

    - by James
    I would like to buy a small embedded system that runs Linux and Asterisk. I would like two FXS ports ( to plug analogue phones into ) and one FXO port ( to plug in to my real line to allow access to the POTS ). I would like it to have a USB port to hold storage for voicemail. I really want it for home use so I would like it to be under £150 ( say $250 ), given that you can buy ADSL routers for around this much can any of them be made to run Linux and Asterisk? I don't want a PC as the power usage would be too high. I am looking for something like this ASDL router but open and able to run Asterisk or another open PBX. At worst I would like a box which had one FXO and two FXS and just made them completely available over IP to a full Asterisk system on a low power Atom system.

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  • Parse and validate Asterisk dialplan before commiting

    - by adaptive
    I recently made a number of changes to my Asterisk dialplan and would like to validate these changes before I commit. I am thinking more from a "write code" - "compile" - "debug" approach. I am very new to Asterisk and am trying to build my dialplan slowly but the server is already in use (by the spouse) so I'd like to minimize interruptions as much as possible. If I can at-least verify that the code is correct, I can then debug in Asterisk as calls are taking place.

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  • Asterisk is hacking itself [duplicate]

    - by Shirker
    This question already has an answer here: How do I deal with a compromised server? 11 answers I've got some strange logs on my asterisk (and there a lot of extensions were tried): chan_sip.c: Failed to authenticate device 6006<sip:[email protected]>;tag=f106f3fe but IP XX.XX.XX.39 is its OWN IP! cat /etc/asterisk/* | grep 6006 returns nothing. asterisk -rv Asterisk 11.4.0 How its possible, that its hacks itself? And how could I trace, where it comes from?

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  • PSTN Trunk TDM400P Install on Asterisk / Trixbox

    - by Jona
    Hey All, I'm trying to get a TDM400P card with FXO module to connect to our PSTN line. The card is correctly detected by Linux: [trixbox1.localdomain asterisk]# lspci 00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I've run setup-pstn which produces the following output trixbox1.localdomain ~]# setup-pstn -------------------------------------------------------------- Detecting PSTN cards and USB PSTN Devices -------------------------------------------------------------- Hardware present! STOPPING ASTERISK Asterisk Stopped STOPPING FOP SERVER FOP Server Stopped Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wct4xxp: [ OK ] wcte12xp: [ OK ] wct1xxp: [ OK ] wcte11xp: [ OK ] wctdm24xxp: [ OK ] opvxa1200: [ OK ] wcfxo: [ OK ] wctdm: [ OK ] wcb4xxp: [ OK ] wctc4xxp: [ OK ] xpp_usb: [ OK ] Running dahdi_cfg: [ OK ] SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP Server Started Chan Extension Context Language MOH Interpret Blocked State pseudo default en default In Service 1 from-pstn en default In Service dahdi_scan returns: dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM400P REV I Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV I location=PCI Bus 00 Slot 10 basechan=1 totchans=4 irq=209 type=analog port=1,FXO port=2,none port=3,none port=4,none And asterisk can see the channel: > trixbox1*CLI> dahdi show channel 1 > Channel: 1LI> File Descriptor: 14 > Span: 11*CLI> Extension: I> Dialing: > noI> Context: from-pstn Caller ID: I> > Calling TON: 0 Caller ID name: > Mailbox: none Destroy: 0LI> InAlarm: > 1LI> Signalling Type: FXS Kewlstart > Radio: 0*CLI> Owner: <None> Real: > <None>> Callwait: <None> Threeway: > <None> Confno: -1LI> Propagated > Conference: -1 Real in conference: 0 > DSP: no1*CLI> Busy Detection: no TDD: > no1*CLI> Relax DTMF: no > Dialing/CallwaitCAS: 0/0 Default law: > ulaw Fax Handled: no Pulse phone: no > DND: no1*CLI> Echo Cancellation: > trixbox1128 taps trixbox1(unless TDM > bridged) currently OFF Actual > Confinfo: Num/0, Mode/0x0000 Actual > Confmute: No > Hookstate (FXS only): Onhook A cat of /etc/asterisk/dahdi.conf shows: [trixbox1.localdomain ~]# cat /etc/asterisk/dahdi-channels.conf ; Autogenerated by /usr/sbin/dahdi_genconf on Tue May 25 17:45:13 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) ;;; line="1 WCTDM/4/0 FXSKS (SWEC: MG2)" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 1 callerid= group= context=default I have configured a "ZAP Trunk (DAHDI compatibility Mode)" with the ZAP identifier 1 and an outbound route, but when ever I try to make an external call via it I get the "All Circuits are busy now, please try your call again later message". I have one outbound route which uses the dial pattern 9|. and the Trunk Zap/1 and one Zap Trunk which uses Zap Identifier (trunk name): 1 and has no Dial Rules. The FXO module is directly connected to our phone line from BT via a BT-RJ11 cable. When running tail -f /var/log/asterisk/full and placing a call I get the following output: [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP RTP TOS bits 184 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP RTP CoS mark 5 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP VRTP TOS bits 136 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP VRTP CoS mark 6 [May 26 11:10:52] WARNING[2661] pbx.c: FONALITY: This thread has already held the conlock, skip locking [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] Macro("SIP/801-b7ce8c28", "user-callerid,SKIPTTL,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:2] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:3] ExecIf("SIP/801-b7ce8c28", "1?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:4] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:5] Set("SIP/801-b7ce8c28", "AMPUSERCIDNAME=Jona") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:6] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:7] Set("SIP/801-b7ce8c28", "AMPUSERCID=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:8] Set("SIP/801-b7ce8c28", "CALLERID(all)="Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:9] Set("SIP/801-b7ce8c28", "REALCALLERIDNUM=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:10] ExecIf("SIP/801-b7ce8c28", "0?Set(CHANNEL(language)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:11] GotoIf("SIP/801-b7ce8c28", "1?continue") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-user-callerid,s,20) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:20] NoOp("SIP/801-b7ce8c28", "Using CallerID "Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:2] Set("SIP/801-b7ce8c28", "_NODEST=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:3] Macro("SIP/801-b7ce8c28", "record-enable,801,OUT,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] GotoIf("SIP/801-b7ce8c28", "1?check") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-record-enable,s,4) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:4] AGI("SIP/801-b7ce8c28", "recordingcheck,20100526-111052,1274868652.1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [May 26 11:10:52] VERBOSE[2858] logger.c: recordingcheck,20100526-111052,1274868652.1: Outbound recording not enabled [May 26 11:10:52] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28>AGI Script recordingcheck completed, returning 0 [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:5] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:4] Macro("SIP/801-b7ce8c28", "dialout-trunk,1,01483890915,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] Set("SIP/801-b7ce8c28", "DIAL_TRUNK=1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:2] GosubIf("SIP/801-b7ce8c28", "0?sub-pincheck,s,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:3] GotoIf("SIP/801-b7ce8c28", "0?disabletrunk,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:4] Set("SIP/801-b7ce8c28", "DIAL_NUMBER=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:5] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=tr") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:6] Set("SIP/801-b7ce8c28", "OUTBOUND_GROUP=OUT_1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:7] GotoIf("SIP/801-b7ce8c28", "1?nomax") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s,9) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:9] GotoIf("SIP/801-b7ce8c28", "0?skipoutcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:10] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:11] Macro("SIP/801-b7ce8c28", "outbound-callerid,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:2] ExecIf("SIP/801-b7ce8c28", "0?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:3] GotoIf("SIP/801-b7ce8c28", "1?normcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-outbound-callerid,s,6) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:6] Set("SIP/801-b7ce8c28", "USEROUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:7] Set("SIP/801-b7ce8c28", "EMERGENCYCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:8] Set("SIP/801-b7ce8c28", "TRUNKOUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:9] GotoIf("SIP/801-b7ce8c28", "1?trunkcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-outbound-callerid,s,12) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:12] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:13] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:14] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:12] ExecIf("SIP/801-b7ce8c28", "0?AGI(fixlocalprefix)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:13] Set("SIP/801-b7ce8c28", "OUTNUM=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:14] Set("SIP/801-b7ce8c28", "custom=DAHDI/1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:15] ExecIf("SIP/801-b7ce8c28", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:16] Macro("SIP/801-b7ce8c28", "dialout-trunk-predial-hook,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:17] GotoIf("SIP/801-b7ce8c28", "0?bypass,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:18] GotoIf("SIP/801-b7ce8c28", "0?customtrunk") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:19] Dial("SIP/801-b7ce8c28", "DAHDI/1/01483890915,300,") in new stack [May 26 11:10:52] WARNING[2858] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [May 26 11:10:52] VERBOSE[2858] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:20] Goto("SIP/801-b7ce8c28", "s-CHANUNAVAIL,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] GotoIf("SIP/801-b7ce8c28", "1?noreport") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:3] NoOp("SIP/801-b7ce8c28", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:5] Macro("SIP/801-b7ce8c28", "outisbusy,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] Playback("SIP/801-b7ce8c28", "all-circuits-busy-now,noanswer") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28> Playing 'all-circuits-busy-now.ulaw' (language 'en') [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [[email protected]:2] Playback("SIP/801-b7ce8c28", "pls-try-call-later,noanswer") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28> Playing 'pls-try-call-later.ulaw' (language 'en') [May 26 11:10:54] WARNING[2661] pbx.c: FONALITY: This thread has already held the conlock, skip locking [May 26 11:10:54] VERBOSE[2858] logger.c: == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/801-b7ce8c28' in macro 'outisbusy' [May 26 11:10:54] VERBOSE[2858] logger.c: == Spawn extension (from-internal, 901483890915, 5) exited non-zero on 'SIP/801-b7ce8c28' [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] Macro("SIP/801-b7ce8c28", "hangupcall") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [[email protected]:1] ResetCDR("SIP/801-b7ce8c28", "vw") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [[email protected]:2] NoCDR("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [[email protected]:3] GotoIf("SIP/801-b7ce8c28", "1?skiprg") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,6) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [[email protected]:6] GotoIf("SIP/801-b7ce8c28", "1?skipblkvm") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,9) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [[email protected]:9] GotoIf("SIP/801-b7ce8c28", "1?theend") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,11) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [[email protected]:11] Hangup("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/801-b7ce8c28' in macro 'hangupcall' [May 26 11:10:55] VERBOSE[2858] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/801-b7ce8c28' I'm guessing I've missed a configuration step somewhere but no idea where, any help greatly appreciated.

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  • Asterisk, IAXModem & Hylafax how-tos?

    - by Brian Postow
    I'm trying to set up Asterisk and IAXModem to send faxes via T38 (Yes, I know I'm swatting a fly with a Buick...) However, since I'm trying to do something so small with a product so large, I'm having trouble finding samples or how-tos that show me how to set this up. I've got all three installed, and I THINK I have my IAXModem config correct. I'm pretty sure that I have Hylafax correct (I've used it with T38Modem) so, I need to know which of the Asterisk samples I need to use, and how to use them. I think I want to use some combination of iax.conf, iaxprov.conf, sip.conf and sip_notify.conf. But I'm not sure where to put them, or what to change... I'm sure that the answer is RTFM, but I'm not sure WHICH M, or where in it to R... thanks. EDIT On a mailing list, someone told me that this actually WON'T WORK because IAX doesn't do T38. So, is there some other way to get Asterisk to work with Hylafax and send T38? I know that Asterisk does T38, the question is how to get the data from Hylafax and back...

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  • Interpreting and using the Asterisk "timing test" command

    - by zigg
    Timing is very important for certain kinds of applications in Asterisk. If DAHDI is the timing source, the dahdi_test command can be used to check the timing provided by the DAHDI kernel module. If dahdi_test returns exclusively measurements above 99.975%, the DAHDI timing source is generally considered good. Since Asterisk 1.6, new timing sources have become available, such as pthread and timerfd. The accuracy of these timing sources seems to be measurable with the Asterisk CLI timing test command: localhost*CLI> timing test Attempting to test a timer with 50 ticks per second. Using the 'timerfd' timing module for this test. It has been 1000 milliseconds, and we got 50 timer ticks My concern is that timing 50 ticks seems to be a considerably less stressful test than dahdi_test's 8192 samples in 8000 ms, particularly since just about every system I've tried it on, virtual or otherwise, can handle it. I can ask timing test to ramp it up to what I think are dahdi_test's standards: localhost*CLI> timing test 1024 Attempting to test a timer with 1024 ticks per second. Using the 'timerfd' timing module for this test. It has been 1000 milliseconds, and we got 1024 timer ticks This will indeed break down a bit depending on the system I'm using, usually with a decrease in timer ticks. But I'm not sure whether this is useful to stress it to this level. Is there authoritative guidance on using and interpreting the timing test command to insure that a given Asterisk system has a timing source that will work well?

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  • Cant connect to asterisk internal database [on hold]

    - by Bilbo
    Im trying to get a PHP script to connect to Asterisks internal mysql database. I tried the to use the standard method for example $con = mysqli_connect("192.168.1.126","root","mysql","asterisk"); However when I log into the asterisk server to access the mysql database all i need it to type "mysql" and im logged in. Im wondering is it possible for my php script to connect to asterisk internal database. The following error is shown: Warning: mysqli_connect(): (HY000/2003): Can't connect to MySQL server on '192.168.1.126' (111) in /var/www/html/project/sipSubScript.php on line 6 Failed to connect to MySQL: Can't connect to MySQL server on '192.168.1.126' (111)

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  • Unable to call through asterisk

    - by sk
    I want to create a voip service. I have installed asterisk-1.4 on a dedicated remotely hosted debian lenny distro. I made a sip.conf and extensions.conf so as to place a call between two sip phones(i am using xlite 3.0) installed in some other Windows PC. Whenever i switch this phones the asterisk console shows that Registration from '"1000"<sip:[email protected]>' failed for '122.168.10.254' - Peer is not supposed to register Where xx.xx.xx.xx is the server's IP. i.e my sip phones are unable to register with the asterisk server. Please help me to place call between two sip phones #sip show peers Name/username Host Dyn Nat ACL Port Status 2000 (Unspecified) D 0 Unmonitored 1000 (Unspecified) D 0 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] # sip show registry Host Username Refresh State Reg.Time # sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 0 active SIP channels

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  • Cant get php script to connect to asterisk internal mysql database

    - by Bilbo
    Im trying to get a PHP script to connect to Asterisks internal sql database. I tried the to use the standard method for example $con = mysqli_connect("192.168.1.126","root","mysql","asterisk"); However when I log into the asterisk server to access the mysql database all i need it to type "mysql" and im logged in. Im wondering is it possible for my php script to connect to asterisk internal database. //edit The following mysql error is shown Warning: mysqli_connect(): (HY000/2003): Can't connect to MySQL server on '192.168.1.126' (111) in /var/www/html/project/sipSubScript.php on line 6 Failed to connect to MySQL: Can't connect to MySQL server on '192.168.1.126' (111)

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  • Understanding Asterisk Server features

    - by Arham Ali Qureshi
    I need to ask few question about Asterisk 1) Does ACL mean by Access Control list here ?If yes than how could i use it? >ip show user 6001 * Name : 6001 Secret : <Set> MD5Secret : <Not set> Context : DLPN_Admin Language : AMA flags : Unknown Transfer mode: open MaxCallBR : 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 2147483647 Callgroup : 1 Pickupgroup : 1 Callerid : "test" <6001> ACL : No Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Sess-Min-SE : 90 secs RTP Engine : asterisk Codec Order : (ulaw:20,gsm:20) Auto-Framing: No 2) What is mean by "Require Call Token" in Asterisk Digium GIU on Create new User Panel 3) Is There any command from where i can get users VOICE MAIL password ? 4) What AMI or CLI command set call recording on or off for user ? and if i want that file to be stored on client computer not on server memory what could i do ?

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  • Asterisk Connection not working

    - by Tamas Ionut
    I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something like 192.168.x.y?? ). However, when I navigated to that url from outside of VirtualBox, that url pointed to nothing. Also I am logged in as [email protected] Should I be logged in as [email protected]? I have also validated the installation as described here and everything went well. I am a complete beginner at Asterisk.

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