search for: gossiptel

Hi, Has anyone got Gossiptel working with Asterisk? - I am having real problems getting it to register - i'm just getting timeout errors. Thanks --ian

Hi, I have Asterisk setup and registered with Gossiptel but i'm only getting 1 way audio. If I call 160 (echo test) or 123 (talking clock), it makes the call but I just get silence. If I call my Gossiptel number from a pstn line, I get gossiptel -> pstn audio but not pstn -> gossiptel audio. I've got ports 5060 and the rtp ports forwarde...

...unning Asterisk 1.2.3 - no additional hardware - everything is going to be running via SIP. To enable inbound and outbound connectivity I have been experimenting with using various accounts provided by Gosspitel, Sipgate, aql and others and have found the most sucessful have been those provided by Gossiptel. Herein lies the problem. I need to register about six incoming lines all provided by Gossiptel - half of them to be active within one context and half within another. I have sucessfully registered all the lines within sip.conf as follows: register => username1:password1:authuser1@sip.gossip...

I have no analog line interfaces on my asterisk system, but I do have two UK 0870 numbers routed to two separate VoIP accounts (one with FWD, one with gossiptel). Asterisk is configured to register with these accounts. I get voice calls through just fine this way. I thought I could get one of these 0870 numbers to route through to rxfax, thus allowing folks to fax me directly. I've set up the spandsp stuff accordingly, and sure enough the CLI shows t...

...changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to Stanaphone, FWD, Gossiptel and PSTN via an X100P. For incoming calls, an 0870 number from CallUK routes to my FWD account, and an 0870 number from Gossiptel routing to my Gossiptel account. Outbound calls all work fine ... I get audio in both directions, no problem. Incoming calls on either 0870 number connect fine, and a...

Hi, If I do a "sip show channels" - I seem to be getting channels left open after calls have ended - any ideas why? I thought at first it was my Sipura SPA-3000 and that Asterisk was not detecting that i've hung up. However, after more testing, it seems to be just on Gossiptel calls - I tried a few of my other sip providers and the channels stay open after the call has ended but then dissapear after about 30 seconds. With Gossiptel calls - the channels just seem to stay open forever (or at least for a long time causing me to get errors about running out of rtp ports). I...

I have a connect to * via H.323/g711 from device A and want to connect to B which want for H.323/g729 h323.conf contains disallow=all allow=alaw allow=g729 but outgoing faststart/TCS contains only g711 (from h323_request(format) i think) and so no codec negotiation and no voice. Howto run up g711/H323 -> * -> g729/H323 PS intel's g729 was used. ast 1.0.3-6 PPS stupid -

This puzzles me. If I start asterisk in the background, and then attach to it to perform some chores, is there a way to detach again without stopping the background process? Entering "stop now" kills both the console attachment as well as the background process. I need to attach to the running asterisk in order to do "init keys" but once I do that, it seems I cannot just

My VoIP provider allows me to have more than one call outbound on the same line simultaneously, for some reason. I am pretty sure that they do not want this to happen, so I'd like instead to limit each line to one call. I do not want the users to have to dial another prefix to go out on another line. Is there any way to add multiple accounts for my _9. extension and have Asterisk

...t; > The errors suggest that while the kernel sources are > installed, the > kernel has not been built. > > Check on the exact procedure for your distribution. > > HTH > > - -- > Ron Wellsted > http://www.wellsted.org.uk > ron@wellsted.org.uk > FWD:519961 Gossiptel:9309811 > N 52.567623, W 2.137621 > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.2.6 (GNU/Linux) > Comment: Using GnuPG with Thunderbird - > http://enigmail.mozdev.org > > iQEVAwUBQi4T10tP/KMNOfRbAQJajAgAso8fLd3qYmBhgfzUBrMDQ8jDE/kWH/4r > jcTiVHcsMxbm1kBxAL5zF9X6rDVpUr...

I installed Asterisk on Gentoo using emerge. At first, emerge tried installing version 0.9 but reading the wiki showed how to get the latest stable. I'm running Gentoo kernel 2.6.11-gentoo-r9. Asterisk seems to be working just fine, but I'm concerned that since I don't have any Digium hardware, I may need a timer source. When I executed emerge zaptel, it installed zaptel 1.0.7

I've got three 7960s running v6 SIP firmware. My Asterisk setup has worked fine with grandstream devices, and basically, we're just upgrading to use nicer phones. Whilst I can make/receive calls from the 7960 to/from gossiptel). When I try to place a call, I get the following Jan 21 11:09:23 NOTICE[19688]: chan_sip.c:7271 handle_request: Failed to authenticate user "30" <sip:30@server.ourdomain.com>;tag=00078599323d000750732f5f-2c61cb72 We're running Asterisk CVS-v1-0-01/18/05-23:43:27 The SIP&l...

I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk -