similar to: Codecs and more analog lines?

Hi! Does * do DNS-lookups when outgoing calls are placed to a different SIP domain? Can I call from <sip:1000@mydomain.com> to <sip:2000@remote_domain.com>? Can * work as a regular SIP proxy in that aspect? Can * handle SIP URI:s that are complete SIP URI:s (sip:user@domain) instead of numbers only? Or should I run a SIP proxy on a different machine to handle pure SIP requests and let

Hi Is it possible (or recommended) to run both Asterisk and say SER on the same physical machine? How about port conflicts? Maybe the easiest way is to change the default SIP port on Asterisk? But how will that work if I register some SIP accounts directly from asterisk (like my SIP provider) but then wanna dial outbound pure SIP calls via my SER... Has anyone got a functional system like this

Hi Is there any work being done on implementing IM/SIMPLE support for SIP on Asterisk? Like a presence server? rdgs, /Staffan Kerker

I have an interesting problem now. I use asterisk to connect to both FWD and a sip provider here in sweden. suddenly, (i know my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try to make a call using this provider. FWD still works fine, and I can call directly towards the GW to POTS without any problems. But, as I call using my providers SER, Asterisk crashes.

Hi I'm running exactly the same setup. Asterisk is running on my FW/NAT/Router with two interfaces. My local phones are situated behind the NAT and connects to the outer interface of the */FW/NAT/Router. * is then connected to my SIP providers (since I'm only using the SIP-part of *, PSTN connection through my SIP-provider). Works fine! rgds, /staffan kerker sweden -----Ursprungligt

Hi I know this is a bit off topic, but still pretty interesting. I'm running Asterisk on my Linux router/NAT/FW connected via cable (1mbit/200kbit) to the internet. Now, I wanna do local QoS implementation. Just very simple to give RTP (UDP) highest priority on my outbound interface. So, whenever I got an ongoing call, the RTP traffic should be handled first and other data (file transfers

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

Hi there, I was wondering if anyone has successfully used Asterisk as a dedicated Analog PSTN gateway to take the place of, for example, a Mediatrix 1204 or an 8 port model? Basically, I am thinking of using a Linksys SPA9000 as the PBX and just need an Analog PSTN gateway for 4 to 8 FXO lines. It does not sound like the Mediatrix 1204 does a very good job and I figure I can build a much more

Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC

Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC

Hi everybody, i've been using rsync for quite some time to sync files between my laptop and my desktop for example i use the following to backup the music library (from Mac to Linux) > rsync -rvzu --exclude=.DS_Store --delete --exclude 'Podcasts/*' > $LOCAL_MUSI $REMOTE_MUSIC >> $LOGFILE An odd thing i always noticed is that some files were always copied, even

Dear package mgcv users, I am using package mgcv to describe presence of a migratory bird species as a function of several variables, including year, day number (i.e. day-of-the-year), duration of survey, latitude and longitude. Thus, the "global model" is: global_model<-gam(present ~ as.factor(year) + s(dayno, k=5) + s(duration, k=5) + s(x, k=5) + s(y, k=5), family =

Processing commands for control at bugs.debian.org: > reassign 745419 src:xen 4.1.4-3+deb7u1 Bug #745419 [xen-utils-4.1] xen-utils-4.1: Pygrub fails to boot from LVM LV when something installed in the volume boot record Bug reassigned from package 'xen-utils-4.1' to 'src:xen'. No longer marked as found in versions xen/4.1.4-3+deb7u1. Ignoring request to alter fixed versions of

Hi folks, I always wondered why there is no "metaflac --show-all-tags", in parallel to --remove-all-tags. Attached you can find a patch for your consideration. Sample output: % metaflac --show-all-tags *.flac 01 Pigs on the wing (Part One).flac:ARTIST=Pink Floyd 01 Pigs on the wing (Part One).flac:TRACKNUMBER=01 01 Pigs on the wing (Part One).flac:ALBUM=Animals 01 Pigs on the wing

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have

> 1) May I know how Speex compared with GIPS codec? It seems that Google, > Yahoo, and Skype are licensing from GIPS. Are there any good benchmarking > or fair comparisons? I think these two emails sum up my opinion about Speex vs. iLBC: http://lists.xiph.org/pipermail/speex-dev/2005-June/003410.html http://lists.xiph.org/pipermail/speex-dev/2005-September/003652.html > 2) In

Hello guys, I need your help related to a mediatrix 1204 configuration. I read some of the messages that you posted in the asterisk users mailing list about the mediatrix 1204 and decided to contact you. I know that the community is not related to Mediatrix devices, but so far I have not found any other group that has work as much as you with them. I bought the mediatrix in Mexico and my provider

I'm having a problem getting outbound trunking to work using asterisk and an external SIP FXO. 7 digit dialing produces the following output: -- Executing Dial("SIP/mitel-fe17", "SIP/5925660@mediatrix-1204") in new stack -- Called 5925660@mediatrix-1204 -- SIP/mediatrix-1204-645e answered SIP/mitel-fe17 -- Attempting native bridge of SIP/mitel-fe17 and

Hi, We are planning to use Speex as the speech codec for a VoIP application. 1) May I know how Speex compared with GIPS codec? It seems that Google, Yahoo, and Skype are licensing from GIPS. Are there any good benchmarking or fair comparisons? 2) In particular, how is the jitter buffer control for Speex in response to intermitent poor connection hiccups? Is it robust enough to smooth out