similar to: usable/affordable usb phone?

Did anybody ever come across an affordable, residential cable/dsl router with support for QoS? The ones I've seen so far (Netgear, D-Link and W-Linx) do not seem to support it. I noticed that even email can damage a G.711 stream on an 128kbit uplink, leave alone file-sharing applications. I understand this is strictly related to *, but nevertheless of interest to many of us. Thilo

Did anybody figure out how to make dial detect a busy on a zaptel channel on a pri interface when using overlap dialing? According to the documentation dial should return to priority n+101, if the called party is found to be busy. I can see a DISCONNECT message with "user busy" coming from the network when I turn on pri debugging, but the dial application does not seem to notice.

[ZapRAS triggering a kernel panic] >> Kernel panic: Aiee, killing interrupt handler! >> In interrupt handler - not syncing >> HDLC Receiver overrun on Channel Tor2/0/2/25 (master: Tor2/0/2/25) > > Hrm, I haven't seen this before. Please contact me off-list and I'll > give you more debugging instructions that may be helpful, as well as > enquire additionally

I see in the changelog that ENUM support has been added.. anyone know what this is? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi Dan, i did it wrong, sorry: curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

Hi, is there a way to enable dtmf detection on zap channels? I am trying to pickup, play a ringtone and the dial out. I.e. exten => s,1,Wait,1 exten => s,1,Answer exten => s,2,Playtones(dial) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => _X,1,StopPlaytones exten => _X,2,Dial,Zap/g8/BYEXTENSION|10

When I was trying to run mutiple E1s over TDMoE, the zaptel would drivers complain about too little memory, whenever I added more than 31 channels. Requesting 62 channels in a dynamic span gave me ... span creation failed: Cannot allocate memory upon loading the zaptel drivers. How would you go about running, 8 or 16 say, E1s over TDMoE? Would you create multiple dynamic spans or just one large

Hi All, I hope someone can give me a hint. We try to reload the asterisk dialplan config using ansible command module. Using this we just trigger asterisk -rx "dialplan reload" Now we want ansibe to fail if there is something wrong in the dialplan. If we put a bad config in extensions.conf dialplan reload prints some warning in Asterisk console but the command "asterisk

Hi everyone, any chance * could be used as a b2bua without forcing the media stream through the same box? I would love to do some computing on incoming calls, do things like setting another callerid and the forward the call to another sip UA - all without any audio traversing the * box. Any ideas? Thanks, Thilo

Hi, I am lost trying to figure out how to enable overlap dialing for calls coming in and coing out on a pri span. DISA looked promising at first, but does not seem to support overlap dialing. Just picking up a call by and trying to dial out does not seem the way to do it either. I tried: [dialincontext] exten => 12341234,1,Goto(dialoutcontext,s,1) [dialoutcontext] exten => s,1,Wait,1

Hi, I'm curious what chip Digium is using in the latest TDM400 cards. Specifically, to my recollection, they used to use the TigerJet 320G, however somewhat recently, Tigerjet was bought out, and now the 320G is no longer produced. Maybe a better question is: is there a way I can take the latest DAHDI source and get a list of supported chipsets from it? Thanks. MCH -------------- next part

Hi, @josh, We are using AGI. It is a very simple perl script. If we need to move to FastAGI we most likely would port it to ARI instead. But I let you know. @john, we using Perl. To see if it is a problem with the perl i had put an "exit 0" just at the first lines so there is no logic done at the AGI. It's only the start up and return from AGI what produces the most of the load.

Hi All, We are using asterisk 16.5 and having an issue with the first re-invite after the call has been established. We can see the call gets up and you see in the logs the bridge type has changed and after that a re-invite is triggered. Is there any possibility to deactivate this kind of reInvite? We have some race conditions while have multiple asterisk in the call flow and the different

Is anybody aware of a database containing the types of nat implementation in todays soho/consumer routers? I think it would make sense for the community to have this database in order to avoid symmetric nats. If one such thing does not exist how about starting this database? A stunclient for linux can be found at http://sourceforge.net/projects/stun/ I can contribute this information for two

Hello, Does anyone know what driver to use with an "Eicon Diva Server 4BRI" card? My modem.conf: -------------- [interfaces] context=remote driver=i4l msn=16453 device => /dev/ttyds01 -------------- I get the following error: -------------------------- == Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated Modem Driver) WARNING[8192]: File chan_modem.c, Line 383

Hi all, So the scenario is: A -> Asterisk -> B after B send back 200 OK Asterisk is answering the call to A. Directly after the Answer Asterisk generates a ReInvite to A and the only difference between the 200 OK sdp and the reInvite sdp are the offered codecs which are forwarded from B to A. Here i do not understand why this could not be done in the 200OK to A? As far as i understood

Hi all, we are working on a A to B basic Call scenario with early media. On that scenario we get a call from a PJSIP endpoint and we place a new call using ARI. On the created channel we receive a 183 Session progress where we have an announcement regarding e.g. the cost of the call (it's important for us to have this announcement to inform our customers about the costs). Using asterisk

I don't have any experience of your problem - but I thought this might help. http://www.hut.fi/Misc/Electronics/circuits/uk_wiring.html <http://www.hut.fi/Misc/Electronics/circuits/uk_wiring.html> The UK (and some of it's former colonies) use a system called 3-wire ringing. Some equipment from overseas requires an adaptor to make it work. I don't know if the LineJack is one

Hi all, I am trying to add a TigerJet usb adapter to my asterisk installation I have the 1.2.4 asterisk bristuffed version, holding two zaphfc (ISDN) cards If I connect the TigerJet adapter to my linux box (Suse 10) i see: Bus 005 Device 001: ID 0000:0000 Bus 004 Device 003: ID 06e6:831c Tiger Jet Network, Inc. Bus 004 Device 001: ID 0000:0000 Bus 003 Device 001: ID 0000:0000 Bus 002 Device