similar to: Error : SIP/2.0 401 Unauthorized

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, 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

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

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 Friend, Can't get hostname environment variable on asterisk dialplan. Help me about how to get hostname environment variable on asterisk dialplan. I have written "export HOSTNAME" in /root/.bash_profile and when i execute "echo $HOSTNAME" then get right hostname but not success through asterisk dialplan. Get environment variable path right value through below

Hello, I'm currently using OpenSER as REGISTER server and Asterisk for the call routing. Do i need the OpenSER nathelper module if i want to offer (mostly) automatic NAT traversal to my users or does Asterisk have the same functionality? It seems that the nathelper module should be able to automatically traverse any NAT as long as the User-Agents use symmetric RTP. Further it is possible (in

Hi all, I want to upgared from asterisk-1.6.2.6 version to asterisk-1.8.0 version. When i execute "make" command for compilation i have seen below errors. In file included from /usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/cdr.h:31 /usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/data.h:233: error: field ?AST_DATA_IPADDR? has incomplete type

No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? Regards, Gus ----- Original Message ----- From: "Chris Hariga" <contact@techselesta.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, October 19, 2003 8:19 PM Subject: [Asterisk-Users] Music on hold... > Hi, > > I need a sound card and mpg123 for music on

Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf, but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA -> High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems

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

Hi all I'm a newbie in asterisk.I ant to know how i ca configure asterisk to work with the nathelper module of openser to fix the nat problem! Thanks in advance! bets regards Serge --------------------------------- Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international.T?l?chargez

Hi guys, I currently have a sip proxy server (sip express router) which registers the sip phones. I need to add voice mail capability, i.e. sip express router will forward all incoming calls to Asterisk if the user does not pick up the call in 15 seconds. The voicemail recording stops when the user hangs up. However, the recording does not end if the user presses the # key, i.e. it is ignoring

Hi Friends, i have installed and configure asterisk-1.8.0. When i have tried asterisk start get below errors and not able to start asterisk. *FD 32767 exceeds the maximum size of ast_fdset!* Thanks in advance. -- Best Regards, Rajnikant Vanza -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, After some research, it seems the only reasonable thing to do in order to get SIP phones behind NAT working reasonably well without fiddling with the DSL router is to have some kind of far end nat traversal mechanism. Is there any way to do this with open source tools? I've seen somewhere that far end nat traversal can be achieved with SER + nathelper does the job... has anybody

Hi List, When we make calls into asterisk with the help of our mobile, landline number, Cisco 79XX series then we didn't able to here any IVR which is playing into asterisk server. But when we dial from SIP softphone then all is working fine and we are able to here the IVR sound files. What is the problem in this case please help me.. -- ----- Thanks and regards Virendra Bhati

Hi, Is there some parameter that I should pay attention to when using externip parameter on sip.conf? I ask this because after using sipsak I've noticed maybe the reason that i don't get voice in one direction could be because at the SDP/SIP messages are references to the 192.168.1.50 (server internal ip) and not the ip from the provider. Is there a way to alter SDP messages to correct

hi i am configuring the SER on centos please can anybody knows abt this if yes please send me the step by spey ser+mysql_rtpproxy+nathelper i want all thies modules thanx and regards prabhakar.G _________________________________________________________________ Get the new Windows Live Messenger! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline

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