similar to: Ports leak

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

This is slightly OT in that it isn't specifically *-related, but I was wondering what the members of the * user community felt about these two subjects. I've been perusing the OpenPBX.org mail list and the current hot topic is the fact that their project has come to a grinding halt. They are concerned that they don't have enough people working on their project. They feel that * has

I am using AMI to Originate a call. I have been able to get the caller id number to be passed through. However, I can't get the name to be passed through. A person I'm working with has a Freeswitch that is able to pass the caller id name and number through for their call. Comparing the Asterisk SIP messages to the Freeswitch SIP messages, I have narrowed the problem down to a single

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, 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 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]

I have finally received and am configuring my new workstations eith the NVS510 graphics cards, and have run into rather a problem. The X server seems to be loading the NOUVEAU driver properly (based on the contents of Xorg.0.log), but I have one X11 application that doesn't work correctly; it runs as though XSynchronized is always True, even though it's explicitly set to False in the

I am trying to use rsync 2.4.6 to try creating a backup using another locally connected hard disk. Thing copy, but sometimes the group is eith numeric (valid). Often the owner is not even shown. This is the command I use for rsync: rsync-2.4.6/rsync -arxlHpogDt /usr/ /backup/usr/ --delete I login as root. Any help would be appreciated. -- Ian Kettleborough ian@idk.com

Hallo list I've been running fedora for donks as my production laptop os, but now I want to set up one of those old laptops to run as a home server running a number of home type vm appliances. I don't want to risk having to tear down and rebuild the setup every 6 months to a year - so, figure centos is my best canditate to run as a stable server. The sort of home type vms I envisage

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

Hello, I'm tring to make a linear discriminant analysis eith lda (MASS). In the value of the resulting object is there information about the value of the centroids of the discriminated group for the discriminant functions? Thank you in advance for your help! Leonardo -- Leonardo Lami lami at faunalia.it www.faunalia.it Via Colombo 3 - 51010 Massa e Cozzile (PT), Italy Tel:

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

Hi all, after moving to parametrized classes, I''ve started to declare them instead of including. Now, I''m trying to understand the differences (internal) between declare and include, and, reading http://docs.puppetlabs.com/guides/parameterized_classes.html again, I found some sentence which confuses me: "Since include wasn’t designed for use with parameterized

Hi I am trying to use icecast to broadcast a realtime conference from freeswitch. But I am having a delay like 20 seconds then I reduced it to 12s. But I don't know if somebody can help me how to reduce it as lower as possible. Thanks Jorge -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Experts. Im working with Asterisk PBXand freeswitch PBX. I have a challenge with FXO card in Asterisk and i could not solve it yet. I hope you could guide me in this regards. When i want route the call to FXO channels, Before the callee answer the phone (pick up phone), The channel is answered with FXO card. How can change this treat so that the callee dont answer the phone, the channel dont

Hi. I am using asterisk 1.8 and everything was working fine when I was at svn 342661. I then upgraded to vrsion 349339 and discovered the following problem -- one of the end points is a freeswitch box which offers a number of codecs, including PCMU. However, when I tried to make a call I got a 488 response and a message "multiple audio streams not supported" in the log. Is this by

The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but

Hello Asterisk community, I'm trying to use BLF with Asterisk Realtime, i've been searching for some info but nothing seems to be clear, can anyone help me eith some ideas to make this work ok? I'va my dialplan with Realtime Thanks in advance -- Saludos Danny Dias SkypeID: danny.dias1

I used to use Wine (from 0.9.x to 1.1.x) on ArchLinux 32 bits to run a Brazilian Dictionary called Houaiss [1], and it worked almost flawlessly. But then I got a new computer and installed Ubuntu 64 bits on it, and now Houaiss' interface is unreadable (same Windows software, not a 64 bits version). I'll provide further information below, and I'd like to know if it's a 64 bits

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