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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 root@localhost. Should I be logged in as root@server? I have also validated the installation as described here and everything went well. I am a complete beginner at Asterisk.

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  • What are some potential uses of Asterisk (PBX) for a "power user"

    - by user26453
    So I read a lot of good things about Asterisk. I am not however looking to run a call center or small business setup. I am still interested what potential uses it has for me as a "power user" and what features I could harness for my communication needs. I'll throw out that I currently use other technologies like Google Voice, Skype, and a cellphone of course. So, what potential uses could Asterisk PBX have for a user like me?

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  • Asterisk Outgoing CDR Logging To Mysql

    - by user3295551
    Trying to utilize the cdr logging (to mysql) using custom fields. The problem I am facing is only when an outbound call is placed, during inbound calls the custom field I am able to log no problem. The reason I am having an issue is because the custom cdr field I need is a unique value for each user on the system. sip.conf ... ... [sales_department](!) type=friend host=dynamic context=SalesAgents disallow=all allow=ulaw allow=alaw qualify=yes qualifyfreq=30 ;; company sales agents: [11](sales_agent) secret=xxxxxx callerid="<...>" [12](sales_agent) secret=xxxxxx callerid="<...>" [13](sales_agent) secret=xxxxxx callerid="<...>" [14](sales_agent) secret=xxxxxx callerid="<...>" extensions.conf [SalesAgents] include => Services ; Outbound calls exten=>_1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@myprovider) ; Inbound calls exten=>100,1,NoOp() same => n,Set(CDR(agent_id)=11) same => n,CELGenUserEvent(Custom Event) same => n,Dial(${11_1},25) same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) same => n(unavail),VoiceMail(11@asterisk) same => n,Hangup() same => n(busy),VoiceMail(11@asterisk) same => n,Hangup() exten=>101,1,NoOp() same => n,Set(CDR(agent_id)=12) same => n,CELGenUserEvent(Custom Event) same => n,Dial(${12_1},25) same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) same => n(unavail),VoiceMail(12@asterisk) same => n,Hangup() same => n(busy),VoiceMail(12@asterisk) same => n,Hangup() ... ... For the inbound section of the dialplan in the above example I am able to insert the custom cdr field (agent_id). But above it you can see for the Oubound section of the dialplan I have been stumped on how I would be able to tell the dialplan which agent_id is making the outbound call. My Question: how to take the agent_id=[11] & agent_id=[12] and agent_id=[13] and agent_id=[14] etc and use that as a custom field for cdr on outbound calls?

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  • What's a good asterisk hosting provider?

    - by MCS
    Does anyone know of a good hosting provider for asterisk? I came across lylix but don't know if they're any good. Of course I could always install asterisk myself on a godaddy vps, but figure it will be easier (and in the case of lylix, cheaper) to have it done for me.

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  • How to stop registration attempts on Asterisk

    - by Travesty3
    The main question: My Asterisk logs are littered with messages like these: [2012-05-29 15:53:49] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:50] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:55] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:55] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:57] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device <sip:[email protected]>;tag=cb23fe53 [2012-05-29 15:53:57] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device <sip:[email protected]>;tag=cb23fe53 [2012-05-29 15:54:02] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:54:03] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 21:20:36] NOTICE[5578] chan_sip.c: Registration from '"55435217"<sip:[email protected]>' failed for '65.218.221.180' - No matching peer found [2012-05-29 21:20:36] NOTICE[5578] chan_sip.c: Registration from '"1731687005"<sip:[email protected]>' failed for '65.218.221.180' - No matching peer found [2012-05-30 01:18:58] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=dEBcOzUysX [2012-05-30 01:18:58] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=9zUari4Mve [2012-05-30 01:19:00] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=sOYgI1ItQn [2012-05-30 01:19:02] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=2EGLTzZSEi [2012-05-30 01:19:04] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=j0JfZoPcur [2012-05-30 01:19:06] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=Ra0DFDKggt [2012-05-30 01:19:08] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=rR7q7aTHEz [2012-05-30 01:19:10] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=VHUMtOpIvU [2012-05-30 01:19:12] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=JxZUzBnPMW I use Asterisk for an automated phone system. The only thing it does is receives incoming calls and executes a Perl script. No outgoing calls, no incoming calls to an actual phone, no phones registered with Asterisk. It seems like there should be an easy way to block all unauthorized registration attempts, but I have struggled with this for a long time. It seems like there should be a more effective way to prevent these attempts from even getting far enough to reach my Asterisk logs. Some setting I could turn on/off that doesn't allow registration attempts at all or something. Is there any way to do this? Also, am I correct in assuming that the "Registration from ..." messages are likely people attempting to get access to my Asterisk server (probably to make calls on my account)? And what's the difference between those messages and the "Sending fake auth rejection ..." messages? Further detail: I know that the "Registration from ..." lines are intruders attempting to get access to my Asterisk server. With Fail2Ban set up, these IPs are banned after 5 attempts (for some reason, one got 6 attempts, but w/e). But I have no idea what the "Sending fake auth rejection ..." messages mean or how to stop these potential intrusion attempts. As far as I can tell, they have never been successful (haven't seen any weird charges on my bills or anything). Here's what I have done: Set up hardware firewall rules as shown below. Here, xx.xx.xx.xx is the IP address of the server, yy.yy.yy.yy is the IP address of our facility, and aa.aa.aa.aa, bb.bb.bb.bb, and cc.cc.cc.cc are the IP addresses that our VoIP provider uses. Theoretically, ports 10000-20000 should only be accessible by those three IPs.+-------+-----------------------------+----------+-----------+--------+-----------------------------+------------------+ | Order | Source Ip | Protocol | Direction | Action | Destination Ip | Destination Port | +-------+-----------------------------+----------+-----------+--------+-----------------------------+------------------+ | 1 | cc.cc.cc.cc/255.255.255.255 | udp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 10000-20000 | | 2 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 80 | | 3 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 2749 | | 4 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 443 | | 5 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 53 | | 6 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 1981 | | 7 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 1991 | | 8 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 2001 | | 9 | yy.yy.yy.yy/255.255.255.255 | udp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 137-138 | | 10 | yy.yy.yy.yy/255.255.255.255 | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 139 | | 11 | yy.yy.yy.yy/255.255.255.255 | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 445 | | 14 | aa.aa.aa.aa/255.255.255.255 | udp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 10000-20000 | | 17 | bb.bb.bb.bb/255.255.255.255 | udp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 10000-20000 | | 18 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 1971 | | 19 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 2739 | | 20 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 1023-1050 | | 21 | any | all | inbound | deny | any on server | 1-65535 | +-------+-----------------------------+----------+-----------+--------+-----------------------------+------------------+ Set up Fail2Ban. This is sort of working, but it's reactive instead of proactive, and doesn't seem to be blocking everything (like the "Sending fake auth rejection ..." messages). Set up rules in sip.conf to deny all except for my VoIP provider. Here is my sip.conf with almost all commented lines removed (to save space). Notice at the bottom is my attempt to deny all except for my VoIP provider:[general] context=default allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=g726 allow=ulaw allow=alaw allow=g726aal2 allow=adpcm allow=slin allow=lpc10 allow=speex allow=g726 insecure=invite alwaysauthreject=yes ;registertimeout=20 registerattempts=0 register = user:pass:[email protected]:5060/700 [mysipprovider] type=peer username=user fromuser=user secret=pass host=sip.mysipprovider.com fromdomain=sip.mysipprovider.com nat=no ;canreinvite=yes qualify=yes context=inbound-mysipprovider disallow=all allow=ulaw allow=alaw allow=gsm insecure=port,invite deny=0.0.0.0/0.0.0.0 permit=aa.aa.aa.aa/255.255.255.255 permit=bb.bb.bb.bb/255.255.255.255 permit=cc.cc.cc.cc/255.255.255.255

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  • What are some potential uses of Asterisk (PBX) for a "power user"

    - by user144182
    So I read a lot of good things about Asterisk. I am not however looking to run a call center or small business setup. I am still interested what potential uses it has for me as a "power user" and what features I could harness for my communication needs. I'll throw out that I currently use other technologies like Google Voice, Skype, and a cellphone of course. So, what potential uses, if any, could Asterisk PBX have for a user like me?

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  • Hardware needed for receiving and recording videcalls in Asterisk

    - by jneves
    I'm planning an Asterisk configuration that should record videocalls and then feed it to an application. From what I've researched, it seems like app_h234m is the way to go (http://www.voip-info.org/wiki/view/Asterisk+app_h324m+compatibility). But it's not clear to me what are the hardware requirements for this. Can someone enlighten me?

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  • What are some potential uses of Asterisk (PBX) for a "power user"

    - by mindless.panda
    So I read a lot of good things about Asterisk. I am not however looking to run a call center or small business setup. I am still interested what potential uses it has for me as a "power user" and what features I could harness for my communication needs. I'll throw out that I currently use other technologies like Google Voice, Skype, and a cellphone of course. So, what potential uses, if any, could Asterisk PBX have for a user like me?

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  • Asterisk Dial() - passing URL to softphone

    - by Giuc
    According to https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial the Dial() application is capable of sending an URL to the extension being called. I suppose there are softphones implementing this, maybe popping up a browser pointing to the given URL - perfect to open up automatically a CRM customer page when receiving a call by identifying his caller id. Do you know of any softphone implementing this functionality?

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  • Asterisk: Forcing a sip peer to connect via ipv6?

    - by growse
    I've got an asterisk server that connects to an upstream provider over a WAN. The upstream provider supports both IPv4 and IPv6 connectivity, and the asterisk server is behind a NAT. When asterisk connects to the upstream sip peer via IPv6, everything works perfectly. The issue I have is that when I configure the asterisk server IPv6 address via DHCPv6, a race condition means that asterisk sometimes ends up attempting to contact the upstream peer via IPv4 (the SIP DNS name has both A and AAAA records). This is because asterisk starts up before the system has a valid IPv6 address. The connection does not work via IPv4 because of the NAT. Is there a way of configuring the peer to specify that it should only be contactable over IPv6? I guess it might be possible to hack together a firewall rule to deny all IPv4 traffic to that IP, but it'd be easier to configure this within asterisk itself.

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  • Asterisk relay between multiple subnets

    - by immoune
    I wonder what's the best way to go when you have phones on multiple networks which are not directly reachable. I have 3 networks 10.3.x.x 10.6.x.x 10.17.x.x My asterisk server resides on the 10.3.0.5 IP. The machines from the 10.6 and 10.17 networks are routed here through VPN tunnels. At this point we don't talk about NAT anywhere on the network just pure routing. Since the 10.3.0.5 PBX has routes back to all the subnet's it has no problem to communicate with softphones/hardphones from these ranges. The problem comes from that Asterisk (as far as I understand) only responsible for the SIP communication part not the Audio/Video transmission which is in P2P fashion done between the devices. So although a client using sipdroid from 10.6.x.x is able to connect to the pbx (10.3.0.5) and dial a bria client on the 10.17.x.x network once the phone rings out and the call establishes no audio will be transmitted simply because it has no way to directly connect there. For this there are multiple solutions described in this text: http://msdn.microsoft.com/en-us/library/ee480411%28v=winembedded.60%29.aspx What I would prefer is to keep these networks segregated as they are now. What would be the best solution? Is it possible to actually relay through all the audio/video information through the Asterisk server? That would be the best in my case, I using Astlinux there which has a lot of other parts. Thanks

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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.

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  • asterisk send mute command to jukebox on incoming call

    - by Jona
    Hi, We're trialling a Asterisk Now server to take over from our ageing PBX system. One of the "nice to have" features would be the ability to pause or lower the volume on the office jukebox if an incoming call is detected. We currently run a linux jukebox which plays music out of the speakers using mpd and can be controlled by the mpc client. We can manually issue the following command to achieve this: mpc volume 20 Does anyone know how to get asterisk to execute this command or some action that we could hook into when a phone call is incoming to specific extensions?

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  • asterisk incoming/miss call notification (to skype)

    - by tike
    My environment: Centos 5.6, Asterisk with freepbx , skype i.e.sends message with parameter skype.sh skype_user message. Now i wanted to send skype notification so that my asterisk server notification are sent to skype rather than email (or to both skype and email). I know, there is voicemail.conf, voicemail_general and vm_email.inc, which has these body created. vm_email.inc emailbody=${VM_NAME},\n\nThere is a new voicemail in mailbox ${VM_MAILBOX} But i dont see where is something like "mail" command. What my thought to do is: instead of saying "mailcmd" pass system ( /path/to/script) and it would simply send message as rest is already configured. Any suggestion where i could run script rather than sending email Or Executing script on every incoming call, so that i could send as notification on every call over the Skype. (however, ultimate goal is to achieve miss call notification or voice mail notification over Skype.)

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  • Grouping extensions in Asterisk

    - by Matt
    We have an asterisk based phone system with multiple clients connected to it. At the moment, all the extension numbers fall in the range 100 - 9999. But, now we have an issue where a particular customer wants to come onto our service and insists on keeping their existing extension list. Is there a way of having extensions 700 for one customer, and also the same number range to another customer. But yet having them belong to a different group? I see there is a concept of groups in asterisk but none of them seem to provide a solution to this.

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  • Grouping extensions in Asterisk

    - by Matt H
    We have an asterisk based phone system with multiple clients connected to it. At the moment, all the extension numbers fall in the range 100 - 9999. But, now we have an issue where a particular customer wants to come onto our service and insists on keeping their existing extension list. Is there a way of having extensions 700 for one customer, and also the same number range to another customer. But yet having them belong to a different group? I see there is a concept of groups in asterisk but none of them seem to provide a solution to this.

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  • Bringing people into an Asterisk conference call

    - by Harley
    I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO is trying to do, but it doesn't work quite properly for me. Here's what happens: 1. Internal person A calls person B 2. Person A presses *0, he is given a dial tone and person B is taken to a conference room 3. Person A calls person C and they can talk, and then person A presses **. 4. Person C is brought to the conference room, but person A is disconnected. In the last step, A should be taken to the conference room as well. Here's the relevant logs, where 230 is person A, 231 is person B, 207 is person C, and 282 is the conference room.

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  • Asterisk Manager API SIPPeers - Permission Denied

    - by Matt H
    I'm wanting to use the asterisk manager api to show the status of all my SIP lines in a PHP web interface. I thought I'd start simple and use telnet to see it working. So I created a user in /etc/asterisk/manager.conf [portal] secret = password read = all,system,call,log,verbose,command,agent,user Then telnet to localhost on port 5038 This is what I get. asterisk ~ # telnet localhost 5038 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Asterisk Call Manager/1.0 Action: login Username: portal Secret: 8u9sdgk Events: off Response: Success Message: Authentication accepted Action: SIPPeers Response: Error Message: Permission denied Why am I getting permission denied? I thought the user has basically full access? Do I need to restart asterisk to make this work? I didn't restart it. On the other hand, I was able to log in which makes me think that the manager.conf has been reloaded as the portal user didn't exist before. Any ideas?

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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 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 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". The FXO module is directly connected to our phone line from BT via a BT-RJ11 cable. I'm guessing I've missed a configuration step somewhere but no idea where, any help greatly appreciated.

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  • How to make possible on Asterisk meetme.conf

    - by kartook
    how can i configure in my Asterisk Server on meetme.conf Details :For conformance bridge extension : virtual Room 1 : Conference Call 567.xxx.xxxx Voice :Enter for conference dial 1 Voice : Enter your conference Pin then press pound my confrance ID: 10935 virtual Room 2 : Conference Call 567.xxx.xxxx Voice :Enter for conference dial 1 Voice : Enter your conference Pin then press pound my confrance ID: 20202 virtual Room 3 : Conference Call 567.xxx.xxxx Voice :Enter for conference dial 1 Voice : Enter your conference Pin then press pound my confrance ID: 30303

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  • Lync 2010, Kamailio, & Trixbox 2.6.23 (Asterisk 1.4)

    - by slashp
    I'm having an issue trying to connect Lync 2010 phone calls with our trixbox PBX. I've gotten to the point where Kamailio seems to be functioning properly and acting as a bridge between TCP traffic (from Lync) & UDP traffic (to the trixbox, as Asterisk 1.4 does not support SIP over TCP). Our Lync box IP: 10.100.10.41 Our Kamailio box IP: 10.100.10.44 Our trixbox IP: 10.100.10.2 The issue I'm running into is as follows when enabling SIP debugging for the Kamailio box: <--- SIP read from 10.100.10.44:5060 ---> PRACK sip:TNECLTSLY01.contoso.com:5068;transport=Tcp;maddr=10.100.10.41 SIP/2.0 FROM: <sip:9121;[email protected];user=phone>;epid=CF2380792B;tag=4852bab430 TO: <sip:[email protected];user=phone>;epid=CF2380792B;tag=3684a6a24e CSEQ: 24 PRACK CALL-ID: 192daae6-00e1-4140-bddd-0394b35d475b MAX-FORWARDS: 70 Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d VIA: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK159fc989 CONTACT: <sip:TNECLTSLY01.contoso.com:5068;transport=Tcp;maddr=10.100.10.41> CONTENT-LENGTH: 0 USER-AGENT: RTCC/4.0.0.0 MediationServer RAck: 1 23 INVITE <-------------> --- (12 headers 0 lines) --- Sending to 10.100.10.44 : 5060 (NAT) <--- Transmitting (NAT) to 10.100.10.44:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d;received=10.100.10.44 Via: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK159fc989 From: <sip:9121;[email protected];user=phone>;epid=CF2380792B;tag=4852bab430 To: <sip:[email protected];user=phone>;epid=CF2380792B;tag=3684a6a24e Call-ID: 192daae6-00e1-4140-bddd-0394b35d475b CSeq: 24 PRACK User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> trixbox1*CLI> <--- SIP read from 10.100.10.44:5060 ---> ACK sip:[email protected];user=phone SIP/2.0 FROM: "John Jones"<sip:9121;[email protected];user=phone>;tag=4852bab430;epid=CF2380792B TO: <sip:[email protected];user=phone>;tag=3684a6a24e;epid=CF2380792B CSEQ: 23 ACK CALL-ID: 192daae6-00e1-4140-bddd-0394b35d475b MAX-FORWARDS: 70 Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d VIA: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK79a21c CONTENT-LENGTH: 0 My SIP trunk on the trixbox looks like this: [from-lync] exten => _+4XXX!,1,Noop(Stripping + from start of number) exten => _+4XXX!,n,Goto(from-internal,${EXTEN:1}) Though I am still having no luck getting the + stripped or the call to go through. Any ideas would be greatly appreciated. Thank you! -slashp

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  • Asterisk: Dropping calls with an "ast_yyerror"

    - by Nick
    I'm having an intermittent issue where asterisk will play our greeting to the caller, and then drop the call instead of making our phones ring. I'm unable to reproduce the problem with any phones I have here, and many callers get through just fine. Some callers though, run into the problem, and I can't find any pattern to it. The bit of information I could find said it was caused by an error in evaluating a dialplan expression. I'm thinking it's this line: exten = START,n,GotoIf($[${FORCE_CLOSED}=TRUE]?CLOSED,1) But I'm not sure what's wrong with it. I see the following error on the console: [Apr 4 16:29:49] WARNING[27038]: ast_expr2.fl:459 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input:=TRUE^ Surrounding Console output: -- Executing [START@AGInbound:1] Answer("IAX2/AtlantaTeliax-10086", "") in new stack -- Executing [START@AGInbound:2] BackGround("IAX2/AtlantaTeliax-10086", 0000_AG_THANK_YOU_FOR_CALLING_AG") in new stack -- Playing '0000_AG_THANK_YOU_FOR_CALLING_AG.slin' (language 'en') [Apr 4 16:29:49] WARNING[27038]: ast_expr2.fl:459 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: =TRUE ^ [Apr 4 16:29:49] WARNING[27038]: ast_expr2.fl:463 ast_yyerror: If you have questions, please refer to doc/tex/channelvariables.tex in the asterisk source. -- Executing [START@AGInbound:3] GotoIf("IAX2/AtlantaTeliax-10086", "?CLOSED,1") in new stack -- Executing [START@AGInbound:4] GotoIfTime("IAX2/AtlantaTeliax-10086", "9:30-17:0|mon-fri|*|*?OPEN,1") in new stack -- Executing [START@AGInbound:5] GotoIfTime("IAX2/AtlantaTeliax-10086", "10:0-18:30|sat|*|*?OPEN,1") in new stack -- Executing [START@AGInbound:6] GotoIfTime("IAX2/AtlantaTeliax-10086", "12:0-17:0|sun|*|*?OPEN,1") in new stack Relevant lines from the dial plan: exten = START,1,Answer() exten = START,n,Background(0000_AG_THANK_YOU_FOR_CALLING_AG) ; See if we're open ; Force Closed if no one's going to be answering exten = START,n,GotoIf($[${FORCE_CLOSED}=TRUE]?CLOSED,1) exten = START,n,GotoIfTime(${AG_WEEKDAY_OPEN_HOUR}:${AG_WEEKDAY_OPEN_MIN}-${AG$ exten = START,n,GotoIfTime(${AG_SATURDAY_OPEN_HOUR}:${AG_SATURDAY_OPEN_MIN}-${$ exten = START,n,GotoIfTime(${AG_SUNDAY_OPEN_HOUR}:${AG_SUNDAY_OPEN_MIN}-${AG_S$ ; ...and we're not. But maybe the time of day has been overridden? exten = START,n,GotoIf($[${OVERRIDE_TIME_OF_DAY}=TRUE]?OPEN,1) ; No override... We're definatly closed. exten = START,n,Goto(CLOSED,1) Any idea what's wrong with the expression? We recently upgraded from 1.4 to 1.6.

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  • asterisk/freeswitch in nat/no-nat setup

    - by pQd
    hi, my current setup - i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured - one on internal sip proxy [for calls between the branches], another - at 3rd party voip providers [ since it's in different countries - those are different providers, but that's irrelevant ]. i was thinking about terminating sip calls on something like asterisk/freeswitch server and having all sip-devices log on just once to such server[s] - mostly to provide things like voicemail, groupcalls, redirections etc. it seems perfectly doable but there is one problem - i cannot find examples how to prepare for nat/no nat. for calls routed to from/to 3rd party voip operator - i'll need handling for nat/stun etc, but for handling of internal calls - i do not want any nat, all traffic should go via vpns to different branches. can you provide me some hints how to configure it? any tutorials? thanks!

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