similar to: Post call QoS in Asterisk 1.4

Ok, so I've asked this question before, and didn't get an answer. So here I go again! Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the

Ok, so I've asked this question before, and didn't get an answer. So here I go again! Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the

I'm planning on setting up some remote agents and before doing so, I did some simple PING tests to measure latency. The average latency I got was 250ms. Does anyone have experience in terms of quality of calls when there is such high latency? Can anyone comment? Thanks, Waldo

Has anyone got/seen Python code to read puppet reports? I added a bunch of these: class PuppetReport(yaml.YAMLObject): yaml_tag = u''!ruby/object:Puppet::Transaction::Report'' def __init__(self, host, logs, metrics, records, time): self.host = host self.logs = logs self.metrics = metrics self.records = records self.time = time However, the Python YAML

How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070803/c6d473ce/attachment.htm

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

Hello, New subscriber here. I noticed that the FTS index is not used in compound searches. Is this expected? Tested in 2.0.0 and 2.0.8: . search BODY "waldo" * SEARCH . OK Search completed (0.000 secs). . SEARCH CHARSET UTF-8 OR SUBJECT "waldo" FROM "waldo" * SEARCH . OK Search completed (1.768 secs). . SEARCH CHARSET UTF-8 OR SUBJECT "waldo" BODY

G''Day I would like to be able to use my VOIP telephone over a saturated ADSL link whilst enjoying optimum audio quality and utilising all of the bandwidth I pay for. It is about this situation that I write. HFSC appears to be the queueing discipline of choice for VOIP. In order for this to work, though, do I have to account for the ATM overhead in the small VOIP packets by defining my

Just curious... Does anyone know if any companies using Asterisk on embedded hardware (out at the customer premisis), such as the Soekris Net4801, to provide VOIP service? Doug.

Can anyone recommend a company that does professional Asterisk recordings for things like IVR, greetings, MOH, announcements, etc? Thanks, Waldo

I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML

I'm looking for an inexpensive way to connect 20 analog phones to asterisk. I could get a bunch of Linksys or Sipura boxes but was wondering if there is a more cost effective way? I came across the Mediatrix 1104 and even the Mediatrix 1124 but that comes out to be almost $100/port. I might as well buy inexpensive IP phone. Does anyone have any suggestions? Thanks, Waldo

So... I'm trying to access CDR(duration) and CDR(billsec) inside h... I keep getting 0. Can I access the CDR function inside a hangup extensions? Asterisk 1.4.13 Thanks, Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible

I'm a little confused. I'd like to build an RPM for Asterisk 1.4. Is it better to modify and use the spec file under redhat/asterisk.spec and run a 'make rpm', OR is it better to build a custom spec file from scratch and use 'rpmbuid -ba' <specfile>? How do people normally do it? The problem I see with a custom spec file is that since the source is all contained

I'm wondering if anyone can recommend one over the other. I'm mostly interested in running open source solutions, so I would prefer if your recommendations are within the open source arena. Basically, I contemplated the idea of using SER as a NAT Helper and possibly as a SIP server for a portion of our user base. We prefer to have Asterisk in the mix because of the additional

I am the tech coordinator in a small public school in rural North Dakota and all of last year I ran Samba on redhat 8.0. I had XP machines that used Samba as the PDC. Worked fine. But now I am getting some XP home machines that cannot log on to a domain. That's okay, because I am determined this year to use solely redhat. In fact, I put the partitions down to 5 gig for Windows and

Hi all, Assuming 1.0.7 is the latest stable version, how/where can I find out the different CVS revisions available and a description of what has been patched/updated in each CVS revision so I can decide whether to leave my 1.0.7 installation as is, or if I need (or think I need) to patch it with a CVS version? Thanks, Waldo

I need to insert/update multiple MySQL columns in a single row with the func_odbc function at the SAME TIME. Someone showed me how to use ARRAY to retrieve multiple values at the same time, but I need to SET multiple values. Can this be done? If not, I will just stick with MySQL, but that's a pain in the ass because the asterisk-addons package has no default rpm spec file for building an

I'm trying to build an Asterisk rpm from the supplied asterisk.spec file. Made numerous changes to get it to work. The architecture of the system I am building on is x86_64. I'd like to build for i686 though. I added a --target i686 to the rpmbuild line in the Makefile, but it looks like it's still requiring 64bit system libraries. When I try to install the rpm on the i686 machine, it