search for: asteriskserv

Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT. Anybody have any ideas? I did notice that when I start gnophone I see iax.c line...

...ISDN ports talking to the AS5300 for the dial-up to complete after authenticating against a RADIUS server. My questions is: can we use only Asterisk to complete/terminate the dial-up connection, removing the AS5300 out of the picture? Current topology to be set-up: Telco --> SS7 --> TE410P-AsteriskServer --> ISDN --> AS5300 --> Internet Ideal topology: Telco --> SS7 --> TE410P-AsteriskServer --> Internet Some posts talk about zapRAS being able to accomplish this, not quite sure though Sounds like possible: http://lists.digium.com/pipermail/asterisk-users/2004-January/026956....

...in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34 schrieb Mark Wiater: > Yves, > > Didn't you say that > >> AsteriskServer: 192.168.1.211 >> SIP-user: 165 > ? > > On 12/21/2016 4:24 AM, Yves wrote: >> . It is sure for 100% that there is no firewall or something else >> mangeling >> in between... another Hardphone works as expected using the same >> Netzworkcable on the same Ne...

So when someone's brute forcing your server is there a way to identify the originating IPs without using a tcpdump? When I get a failed auth on the console it shows 'account at asteriskserver' then tag=as25ca5023 (or some random string, though it's a bit odd as alwaysauthreject = yes is on in sip.conf). Anyway, the logs don't show anything more useful either. Is there something obvious I'm missing? Cranking up verbosity on the console doesn't seem to do anything...

I sent this last night but it never showed up in the thread list so I'm trying again. Please pardon me if it duplicates. So I've been banging my head against the rack on this one and am now turning to the group for help. I'm in the process of bringing five Asterisk servers (all originally built from source code by myself) from various versions (1.6.2.x,11.6-cert13, and 13.1-cert2) up

...load: http://www.sokol-associates.com/Downloads/IaxPhone.zip Reference & Support Page: http://www.sokol-associates.com/IaxPhone.htm Features: - Works correctly for both inbound and outbound calls! - Registers With Multiple Servers - 4 Line Appearances - Direct IAX URL Dialing (user:password@asteriskserver.com/1123@users) - 20 Speed Dials - Native IAX Blind Transfer (no more wasted pound key) - Drag/Drop Transfer (right-click & drag a line to a Speed Dial to transfer) - Last number redial - Message Waiting Indicator (Native IAX, no Manager configuration reqired) - Message Waiting Count * - Regi...

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

...yves >> > I am a bit confused: is your problematic phone's IP 192.168.0.13 > (what the error log is reporting below) or 192.168.1.13? > >> Am 21.12.2016 um 13:34 schrieb Mark Wiater: >> >> Yves, >> >> Didn't you say that >> >> AsteriskServer: 192.168.1.211 >> SIP-user: 165 >> >> ? >> >> On 12/21/2016 4:24 AM, Yves wrote: >> >> . It is sure for 100% that there is no firewall or something else mangeling >> in between... another Hardphone works as expected using the same >> Netzworkc...

...asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied Apr 29 22:25:25 WARNING[16253]: loader.c:554 load_modules: Loading module codec_g729a.so failed! Ouch ... error while writing audio data: : Broken pipe uname -a (Updated the hostname from the output.) Linux asteriskserver.XXXXXX.XXX 2.6.16-1.2096_FC5 #1 Wed Apr 19 05:14:36 EDT 2006 i686 i686 i386 GNU/Linux I re-downloaded the codec and attempted the i686 and i586 version wiht no luck. md5sum codec_g729a.so 92b64cc5be4a3e622c91357b116d99e3 codec_g729a.so Thanks -Jason -- ------------------------------...

Hi list, how does one get a SNOM 200 MWI to work with * ?? When I press the MWI button it doesn't connect with voice mail on my * box. thanks

...nder the terms of the Apache License 2.0. Here is the Changelog: Bug * [AJ-47] - AGI does not support multi line data * [AJ-51] - Problems with non-english locales * [AJ-52] - Fix shutdown when using the live api Improvement * [AJ-41] - Add ability to get ManagerConnection from AsteriskServer * [AJ-49] - Support socket read timeout New Feature * [AJ-35] - Support timestamp property on manager events * [AJ-42] - Add support QueueSummary action to Queue manager interface * [AJ-44] - Support PauseMonitor and UnpauseMonitor actions * [AJ-45] - Support...

Dear All, I would like to ask please about how to fix the problem of sending fake ring back tone by asteriskserver when trying to make a call from an extension registered on asterisk to any PSTN number...I made some comparaison between calls made through Asterisk server that generate a fake ring back tone during dialing and other sip proxies that does not generate such tone...I noticed that Asterisk server i...

I want to collocate an * box somewhere, where better than where voicepulse chose to put their servers? They probably did their homework and selected someplace where good handoff to the pstn can be found, right/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050520/9f5975b8/attachment.htm

Hello List - I've done some reading on voip-info regarding hardware requirements for an Asterisk server; but I haven't been able to find anyone doing what we plan to, so I am hoping you can assist. We are looking to provide a voice mail only Asterisk solution for approx. 100 homeless people, a customer of ours is planning to provide the service. The Asterisk service will reside in our

Does anyone have some decent Nagios scripts out there that do more than monitor the proc itself? Rather than reinvite the wheel, figured I'd ask. I already saw the one on the wiki. Matt

All: My company has an existing ESI IVX E-class system with 45 phones. I can add one more card, to expand it another 6 phones, but it's $8000, and then the system will have to be replaced. I have the Asterisk server up and running, with 2 sip lines from the local phone service. (Thanks to you guys, it is working great!). I'm pretty sure this is the way the company will move, and

Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 10000-20000. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra