[asterisk-users] ast_debug messages not showing up
Hi, I'm running Asterisk 1.8.7.1 on Gentoo. I set `core set debug 9` but don't see any debug messages on the console. I do get the verbose messages from ast_verbose. Is there something I need to configure to see these messages? Regards, Yahya -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk w/ Nokia e Series Handsets
On Mon, May 11, 2009 at 11:24:36AM -0400, Cory Andrews wrote: Anyone using Nokia E Series handsets with Asterisk? I'm trying to deploy some e71's and am having an issue. I can get a single handset working, but when I try to create a SIP profile on the second phone, it won't allow me to save the profile, saying that devices in the same realm must have identical username and password. Anyone have a workaround for this to add a second Nokia phone under the Asterisk realm with a different userid and PW? I have an E71 and E61i working with asterisk in my office network without problems. I did have a different problem when I used a second SIP profile on the E61i with my home asterisk server and it would not register. Turned out I could not have the same realm name with the same username and password on the two profiles. Changing the realm name on my home profile and asterisk fixed the problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for good IAX ATA
On Wed, Apr 22, 2009 at 09:20:05PM +, Jeff LaCoursiere wrote: I have been wondering - if you ran your SIP traffic over VPN tunnels, what would the state think of that? They obviously won't be able to inspect the data to see what is flowing through the tunnel. Do thye also restrict encrypted traffic? They can't see the data due to encryption, but they can observe packets' traffic patterns, and match them to typical usage by various protocols. Bitek ( http://www.bitek.com/index.php?option=com_contentview=articleid=74Itemid=60 ) is one company that caters to state ISPs and I've heard their technology can guess if voip is being used inside encrypted tunnels. VPNs are not allowed on residential connections, but it's not really enforced, and it does work well with voip. In any case, if an ISP can block 95% of VoIP usage, it's good enough for them. It's probably not worth the effort to block the remaining 5% tech savvy people. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for good IAX ATA
On Fri, Apr 10, 2009 at 09:15:50PM -0400, Marc Charbonneau wrote: Didn't try them myself, but I found those 2 - http://x100p.com/products/FXS.php - http://www.atcom.cn/En_products_AG188N.html I bought an Atcom AG188N ATA recently and it supports both SIP and IAX2, has a built in router, supports g711, g729, g726, and iLBC and can be configured via a web interface, or remote provisioning. I like it much better than the Digium IAXy. Configuring the IAXy is a pain, and the fact that the server address can be specified by IP address only is particularly annoying. The included power supply was 110 volt only making it bothersome for travel usage. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for good IAX ATA
In places where SIP won't work for some reason, I register the phone to asterisk on my laptop which then converts the SIP channel to IAX. How did you do this? Were you using Wi-Fi to talk to the laptop (which was using Wi-Fi to talk to the world?) Yes, that's how I do it. A native Symbian IAX client for the Nokia which would use Wi-Fi (or packet data connection!) would be the way forward. Ignore SIP entirely. The mobile networks don't like you running VoIP over their data streams though, however they don't seem to block it, but it is mentioned in the TC's - at least for the UK networks I've used. Yes, that would be better, but a SIP to IAX adapter on the phone would be less work. The SIP client just needs to register to localhost. No need to develop a user interface for a separate IAX client. Btw, there are other options such as Fring, which I believe uses a proprietary protocol from the cellphone to Fring servers, which is then converted to SIP, Skype, Yahoo etc. I live in the middle east, and the state run ISPs block SIP using deep packet inspection technology. Moreover mobile data packages are so expensive that it's cheaper to make a cellular call. Here's an amusing thought: My Nokia E90 has a SIP client built-in, and it doesn't support the GSM codec - only G711 and G729! I once used it via Wi-Fi when in a conference call - the call lasted 45 minutes and it nearly exhusted the battery and the phone was very hot by the end of it (I was using a headset to the phone!) Good for quick calls, not yet for daily use I reckon. I noticed no such issues on my E61i. Maybe you had a weak wifi signal and the phone ramped up its radio Tx power? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for good IAX ATA
On Sat, Apr 11, 2009 at 11:52:51AM -0400, John Rogers wrote: Most ATAs I've seen are primarily SOC (System On Chip) implementations. I've never really taken one apart, but perhaps now is a good time. I recently also purchased a QuickPhones QA-342 wifi rechargeable handset: http://www.voipsupply.com/quickphones-qa-342-wifi-sip-phone?utm_source=quickphones-wifi-catutm_medium=banner I was curious to try this out, even though I knew it was SIP only. In the office, it works GREAT - LONG battery life, good reception, but no IAX support. A good IAX ATA and IAX protocol stack in a phone like the QA-342 would be a hands down winner all around. Immagine, being able to roam anywhere with a device like this or ATA and not having to fuss with SIP/NAT. I wish I had knowledge on building embedded devices, else I'd build my own by now... I prefer the Nokia E-series wifi enabled cell phones that have a SIP client builtin. I have an E61i and it works great in a wireless hotspot. In places where SIP won't work for some reason, I register the phone to asterisk on my laptop which then converts the SIP channel to IAX. An idea is to write SIP to IAX conversion server software for the phone itself using free libraries which I imagine should be easier that building your own embedded hardware. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax2 not registering at startup, works on reload
On Tue, Mar 31, 2009 at 10:27:45AM +0100, Steve Davies wrote: Most commonly, if DNS is not ready to resolve a hostname, IAX can stall and/or fail to register. DNS was the cause. Replacing the hostname with its IP address fixed it. Thanks! -Yahya ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iax2 not registering at startup, works on reload
I'm running asterisk on Ubuntu 8.10. I have two 'register' lines in iax.conf for registering with two remote servers. However only the first one registers at system startup. I always have to issue an 'iax2 reload' command before * registers with the second remote host. This only happens at system boot. If I stop and start asterisk using /etc/init.d/asterisk script the problem does not occur. Any ideas? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
