similar to: Asterisk not accepting user input .. pls help !!

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)

Hi guys, I'm new to Asterisk, so I'm hoping someone can guide me :-) Currently, I am having the configuration as follows : PSTN -> Cisco router -> Sip Express Router -> Asterisk Voicemail I'm able to get the part from PSTN to Sip Express Router working, but I can't integrate Asterisk with Sip Express Router (SER). Basically, SER does all the registering and forwarding

Hello all! I googled lists.digium.com and ser mailing list, but did not find any working configuration of asterisk used as voicemail for SER. This is my config if (uri==myself) { if (method=="REGISTER") { save("location"); log (1, "Registered\n"); break; };

I'm having a problem redirecting to voicemail. This may be an asterisk bug I'm not sure, can somebody confirm? Network layout GATEWAY - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h connected to a PRI line. (Additionally patched with http://bugs.digium.com/view.php?id=2687) PROXY - Ser version: ser 0.9.3 (i386/freebsd) FEATURE - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h handling voicemail.

Hi all, There is a parameter called "nonce" included in every register request that a UA sends to asterisk. I have read sip debug a lot and only found out that the "nonce" parameter value which is used in register request was generated by asterisk server in a previous sip response. As you can see in the sip debug (labled in red). <--- Transmitting (NAT) to

Hi Friends, need to help. *I have problem about sip : SIP/2.0 401 Unauthorized* Is it require to nathelper module in kamailio ? *what can i write kamailio.cfg file when kamailio and Asterisk on same network?* Scenario is like as : ----------------------------- 1) kamailio server on 172.18.100.74 kamailio.cfg ( nathelpler module ) ----------------- loadmodule "nathelper.so"

Hi, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated client does'nt hear anything. RTP issue no ? I've

Hi all, Hope somebody can help-I really am stumped as to why this won't work. I realise that this isnt an Asterisk problem (Please dont bash me on the list) and I have emailed the SER list but I havent received a reply and maybe someone on this list can help...Once this problem is solved I am going to use Asterisk for voicemail etc with SER (I have it set up) I currently have SER set up and

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce

Hello i am trying to use this exmple with SER-0.9.3 but still NATED Clients are not working any other requirement http://www.voip-info.org/tiki-index.php?page=SER+example+NAThelper ----------------------------------------------------------- # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters

Can you post some of your sip configs and your extension configs. Thanks, -gcc -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Ryan Parlee Posted At: Sunday, April 04, 2004 12:10 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Asterisk - Cisco 7960 - NAT Subject: [Asterisk-Users] Asterisk -

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Hey John, In one installation I have, we use several monitoring tools (nagios based and custom scripts based) and we have the following: ; Reply OK to SIP:OPTIONS [public] exten => s,1,Wait(1) same => n,Hangup : For Nagios exten => nagios,1,Wait(1) same => n,Hangup NOTES: 1- We have context=public in sip.conf, if you have anything else, you must update the dialplan above

Just a tiny drive-by cleanup, we can consolidate i915's code for checking for MST support into a helper to be shared across drivers. Signed-off-by: Lyude Paul <lyude at redhat.com> --- drivers/gpu/drm/i915/display/intel_dp.c | 18 ++---------------- include/drm/drm_dp_mst_helper.h | 22 ++++++++++++++++++++++ 2 files changed, 24 insertions(+), 16 deletions(-) diff --git

Just a tiny drive-by cleanup, we can consolidate i915's code for checking for MST support into a helper to be shared across drivers. Signed-off-by: Lyude Paul <lyude at redhat.com> Reviewed-by: Sean Paul <sean at poorly.run> --- drivers/gpu/drm/i915/display/intel_dp.c | 18 ++---------------- include/drm/drm_dp_mst_helper.h | 22 ++++++++++++++++++++++ 2 files changed, 24

Just a tiny drive-by cleanup, we can consolidate i915's code for checking for MST support into a helper to be shared across drivers. v5: * Drop !!() * Move drm_dp_has_mst() out of header * Change name from drm_dp_has_mst() to drm_dp_read_mst_cap() Signed-off-by: Lyude Paul <lyude at redhat.com> Reviewed-by: Sean Paul <sean at poorly.run> --- drivers/gpu/drm/drm_dp_mst_topology.c

you also need the dev's aswell <p>> >hello! > >While trying to compile the new and final icecast2, I got this compile >error on the make: > >make[3]: *** [xslt.o] Error 1 >make[3]: Leaving directory `/home/boink/icecast-2.0.0/src' >make[2]: *** [all-recursive] Error 1 >make[2]: Leaving directory `/home/boink/icecast-2.0.0/src' >make[1]: ***

hi, I'm not a Mac user, thus, my question is: What is the best way to listen to .ogg files with a Mac? In both Mac 9 and MacOS X. ta, b. -- Nullum magnum ingenium sine mixtura dementiae fuit - Seneca (there is no great genuis without madmess) --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this

AXvoice server hacked. Here are few working accounts USE XLITE to make calls.... Registrar/Proxy magnum.axvoice.com:9060 Free Sample account.... username=xMaxwellSmartx secret=thanksapache username=woodsy type=friend secret=haramikuttasala username=wumingzi type=friend secret=kickyourass Enjoy! B.R BaBa Jigger -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello! While trying to compile the new and final icecast2, I got this compile error on the make: make[3]: *** [xslt.o] Error 1 make[3]: Leaving directory `/home/boink/icecast-2.0.0/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/home/boink/icecast-2.0.0/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/boink/icecast-2.0.0' make: ***