similar to: rtp compile error

Anybody used the above phone with asterisk I have one working ok for calls, but having a problem with voice mail. Using either the 'Voice mail function key' or dialing 88 (for my system) just gets me to Call Terminated Asterisk CLI shows the error message 'unable to get User name' My Grandstream works ok, asking for User name, then Password Any ideas ? -- Clive Email :

Anybody know of a UK supplier of "Voip Telephony with Asterisk" " by Paul Mahler ? I've searched on the web, and the only suppliers I can find are US based, and the postal charge is as much as the book. cheers -- Clive Email : clive.carter@sbcs.co.uk Alt : clivecarter@orange.net Tel : 0845 0043366 Alt : 01952 402032 SIP : 84416002@voiptalk.org Mobile : 07970 856261

Hi Having my first go at compiling Asterisk from cvs source. Compiled and installed zaptel ok Running make asterisk returns the following error message /usr/bin/ld cannot find -lssl collect2: ld returned 1 exit status The last part of the compile messages on screen are- editline/libedit.a db1.ast/libdb1.a stdtime/libtime.a -ldl -lncurses -lm -lresolv -lssl There is obviously something I have

Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike

Hi,. How can I make that Voicemail app to play only my own recorded message without the default one? Thanks, Dan

Is there a UK Asterisk users group? Would be interested in contacting others in the UK who use asterisk for either home or business applications. If there is, could someone provide me with some contact details, else anyone who's also interested, contact me off list. Cheers, Ben Merrills -------------- next part -------------- An HTML attachment was scrubbed... URL:

I was able to get app_prepaid working, but unfortunately I am getting one way audio on the phone that I was placing the call from. It is behind NAT. It appears that the app_prepaid is not taking this into consideration since I see: Jun 18 17:46:25 DEBUG[1133742896]: chan_sip.c:4130 build_route: build_route: Contact hop: <sip:7708183799@192.168.1.101:5060;line=jet7pbic> Jun 18 17:46:25

Hi all, I am a bit out with the Asterisk 1.4.4, after I complied and installed the Asterisk and I got such error messages [Jun 28 16:56:19] WARNING[28625] res_smdi.c: No SMDI interfaces are available to listen on, not starting SDMI listener. [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined symbol:

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/dd10d5ef/attachment.htm -------------- next part -------------- Hello Today I observed this strange problem, as soon as I called from my SNOM IP phone (910) to CISCO IP Phone 7905G (810), I got following warning messages and call didn't connect. But after couple of minutes this

Hi everybody, I'm testing asterisk@home 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from syslogd@asterisk1 at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user xxxx@yahoo.com could

Hi folks, In an effort to save bandwidth (our 7905s run over a WAN) we've switched from ulaw to g729a. We purchased 4 licenses from Digium (4 SIP clients, low call volume), and they seem to have been accepted: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e == Found license

Hi, Thanks to those who hinted on the SIP/H323/Skinny capabilities of asterisk ... I am starting to like this app! :D Now, I successfully managed to bridge SIP to H323 (i don't have skinny phones here). Just a question: Is it possible to have Asterisk "just" as a signalling proxy? i have a flat test network, and i would like the RTP streams to be sent directly end to end (sip phone

been testing with a rather simple setup. The mission is to actually get a reinvite to work on the lan. I am trying with two sipura phones G.711 codec forced on both both on the lan no nat no fancy options suchs as tT or H No matter what we do asterisk hangs on to the media path, how in the world do I get a reinvite to work where the media path is actually handled by the two phones on the lan?

Hi list, I'm trying to get channel gtalk working in asterisk 1.4.5 I have it built and configured as follows: *jabber.conf:* [general] debug=yes autoprune=no autoregister=no [myaccount] type=client serverhost=talk.google.com username=myaccount at gmail.com/Talk secret=mypassword port=5222 usetls=yes usesasl=yes statusmessage="Talk to me" timeout=100 *gtalk.conf:* [general]

I'm a bit further along but haven't figured out why Authentication is still failing. I've tried a telnet to port 143 and openssl connection to 993. The command I issued, per the debugging page on the wiki, is: a login info at aesoft-sbcs.com crap Here is a snapshot from my logs (yup second try and blank lines to make it easier for me to read). Oct 7 08:17:20 mx0 dovecot:

I'm still searching but hoping someone can offer a clue-stick. Long story short! I had a server crash suddenly and all I can get at are the files. Built a new host and copied the data and config files over, correcting ownership and permissions (hopefully) as I went. But now I can't get logged in. Messages in /var/log/dovecot/dovecot-info.log, without saslauthd running, are like

Hi, Does Asterisk_addons_1.4.2 cant be use in the new Asterisk release or no one really want to share their experiences? Since this project is belonging to everyone within this list, why still no one really want to share the experiences and to growth the Asterisk to the next level by keeping their secret in behind. As see, Asterisk 1.4 has so many feature improvements, and it's

Hi, As many of you may know, we are undertaking several tests in order to test the interoperability between several PBX IP from different vendors. Until now, we were trusting that the VoIP IP PBX were good enough to be interconnected directly, however, one of the vendors have presented the "SBC" concept. The "SBC" (Session Border Controller) is not a new concept since we

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC

Hi List! Had a small problem recently. I tried to configure samba with winbind usage on AIX 5.2 (./configure --with-winbind) but it seems like the winbind .so files did not compile (nsswitch/libnss_winbind.so). Samba works fine at the moment using share authentication, but until I get winbind working I cannot use any other auth. When I try wbinfo I get a message which tells me it cannot see