search for: bagdasarian

Hello, I'd like to use the AMI interface to originate a call to a context in a dialplan, and handoff the dial control to the context. Whenever I execute the below action, the recipient does ring, but when I answer it dials the recipient again. I believe this is because once answered the system is going to execute the Context/Exten/Prio in the Originate action? Action: Originate Channel:

Hello, We have a Kamailio SIP Proxy in front of our Asterisk cluster for incoming calls from our carrier. The sip.conf looks like this: [kamailio1] type=friend host=10.0.0.1 context=incoming disallow=all allow=alaw All calls hit the incoming extension. In the extensions.conf we have multiple extensions configured, but now I have to add one which uses the special h extension to perform a CURL

Hello, Does anyone know how frequent segment faults occur in the current LTS release (version 11) and in the future LTS release (version 13)? We are currently using 1.6, which frequently throws unexplained segment faults, that's why we are considering to upgrade to the latest LTS version. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such

Hello, I'm researching the possibilities of multiple communication platforms like Asterisk and FreeSwitch for handling a dynamic sequence of applications to execute, like Playback, Read, etc. This only applies to originating a call from an external application by using the AMI Manager and the Originate action. I need to know the following: 1) Does the Originate action support multiple

Hello, We're running Asterisk 1.8.14.1 and our carrier requires us to send a 200 OK for OPTIONS request in order for them to keep sending traffic to our endpoints. Asterisk is currently replying with 404 messages, and their SBC only accepts 200 OK responses. How do I configure asterisk to reply with 200 OK without changing any source code? Regards, Grant -------------- next part

On Tue, Feb 17, 2015 at 5:14 AM, Grant Bagdasarian <gb at cm.nl> wrote: > > Hello, > > > > We?re running Asterisk 1.8.14.1 and our carrier requires us to send a 200 > OK for OPTIONS request in order for them to keep sending traffic to our > endpoints. > > Asterisk is currently replying with 404 messages, and the...

Hello, Is it possible to log the raw signaling of Dahdi channels to a log file? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150320/31b11a27/attachment.html>

Hello, I'm noticing strange behavior in one of our Asterisk nodes where the ACK is always sent to the proxy, but RR is not enabled for calls. The proxy drops the ACK. I'm using the AMI interface to originate a call: Action: login Username: myusername Secret: mypassword Events: on Action: Originate Channel: SIP/<SOMENUMBER>@proxy1 CallerID: <SOMENUMBER> Application: Playback

Hello, I'm trying to execute a stored procedure on a MSSQL Server from the dial plan, but it's not working. I'm getting the following error: Unable to execute query.... Asterisk has been compiled with UnixODBC, and I've done the necessary configurations in func_odbc, res_odbc and odbc.ini. Has anyone done this before with success? -------------- next part -------------- An HTML

Hello, I'm new to Echo Cancellation and I was wondering how it is handled/works on pure VoIP networks using Asterisk? I did some research on the internet about EC on VoIP networks, but I can't really put a grasp on it. We currently have some Echo Cancellation chips on our Digium cards, but are planning to move to a full VoIP network based on Asterisk. So no more ISDN in the voice path.

Hello, Could someone explain to me what this means? asterisk[30269]: segfault at 0000000000000008 rip 00002aaac8b388f2 rsp 0000000040a75910 error 4 Also, would this segfault crash the whole Asterisk process or will Asterisk continue to run? Is it possible this would affect/disconnect "SOME" DAHDI channels, but not all? At this point, upgrading is not an option, even though I agree we

Hello, I have a debian server(7.2) which has a mount to a windows shared directory. The share is configured under a specific user, let's say bobsshare. I also have a local user on the debian server named alice, but alice cannot write to the mounted directory. What do I need to do to give alice write permissions on the mounted directory? smbclient --version Version 3.6.6 fstab

...irect_media=no (Matthew Jordan) > 5. Re: Asterisk 13 : SILK codec ? (Matthew Jordan) > 6. Problems playing an audio file over an intercom/paging system > (Tech Support) > 7. Asterisk on OpenWrt (first time user) (Sebastian Kemper) > 8. Dahdi ISDN logging (Grant Bagdasarian) > 9. Re: Dahdi ISDN logging (Tony Mountifield) > 10. UNREACHABLE peer (thufir) > 11. Re: UNREACHABLE peer (dotnetdub) > 12. Re: UNREACHABLE peer (thufir) > 13. Re: UNREACHABLE peer (thufir) > 14. Re: UNREACHABLE peer (thufir) > 15. Re: Caller ID Names (Jo...

Hello, We have a few Asterisk 1.8.14.1 boxes which occasionally suffer from audio being lost in a bridged call. So, the inbound and outbound channels are talking to each other for a few minutes, no problems so far, and then suddenly they can't hear each other anymore. These calls are recorded and in the recording you still hear both ends, while they don't hear each other. I tried

Hello, Our supplier requires the From header of a SIP INVITE to contain certain data so the call is placed with a private caller id. It needs to be like this: From: <sip:anonymous at anonymous.invalid;user=phone>;tag=123455667 How do I configure Asterisk to dial anonymously? Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm trying to change the voice during a spoken text: exten => _X.,1,Answer exten => _X.,n,MRCPSynth("Hello, my name is Daniel. I have a Dutch companion. ###\voice=Xander\ Hallo, mijn naam is Xander.",p=default&l=en-GB) exten => _X.,n,Verbose(1, ${SYNTHSTATUS}) exten => _X.,n,Hangup This exact same text, with escape sequence, works perfectly in the demo

Hello, Using Ubuntu Server 12.04 and Asterisk 11.2.1. I'm getting the following error when trying to start asterisk: (Syslog) kernel: [ 1032.713864] asterisk[26918] trap invalid opcode ip:7fc272923076 sp:7fff928cf1b0 error:0 in codec_ilbc.so[7fc272921000+e000] We were running Asterisk on a physical box, but moved it to a virtual environment. That went fine. Asterisk started normally and