similar to: Asterisk 1.6.2.12 segfault

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, 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

Hi list, I'm trying to create an asterisk 1.8 rpm with SRTP. I found mention of a libsrtp rpm, <http://qutecom.ipex.cz/RPMS/srtp-1.4.4-1.i386.rpm > in these instructions, <http://www.voip-info.org/wiki/view/Asterisk+SRTP> but it is unreachable (by me, anyway). The libSRTP source is here, <http://srtp.sourceforge.net/download.html>. Has this already been packaged for

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 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 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, 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

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 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

what is the difference between FreeSwitch and Asterisk , whitch one is more scalable and reliable? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. Regards

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

Hi I am trying to deploy freeswitch with Digium TE121 card for my office setup, but it is continuously showing Signaling is up and channels are down except D channel. Our Architecture is like We have freeswitch installed with libpri1.4 and Dahdi. I am from India and here we are having E1 trunk. Dahdi Configuration is cat system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 7

Hi all, Today I got problem below and my domU become unresponsive and I should restart the pc to make it running properly again. [ 240.172092] INFO: task kworker/u:0:5 blocked for more than 120 seconds. [ 240.172110] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [ 240.172376] INFO: task jbd2/xvda1-8:153 blocked for more than 120 seconds. [ 240.172388]

Hi all, I'm new to VoIP, now we have a project that needs a PBX with client APPs. In our team we have argument for choosing PBX. By so far, we have following candidates: A: Open source 1) Asterisk PBX (http://www.asterisk.org) (with longest history that almost every one knows it, now the last version using the PJSIP stack) 2) FreeSwitch (http://www.freeswitch.org) (A lot people

https://bugs.freedesktop.org/show_bug.cgi?id=83168 Priority: medium Bug ID: 83168 Assignee: nouveau at lists.freedesktop.org Summary: No display after suspend to RAM QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified OS: Linux (All) Reporter: nsajko at gmail.com

Hi Please check out this product http://www.sangoma.com/products/hardware_products/data_networking/a301.html Does it work on Asterisk or Freeswitch ? Do Telcos provide an E3 connection ? One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does