similar to: No subject

Dear all, I have enabled stun module and configured it in asterisk , but asterisk not using stun returned public ip address for any of the sip requests going out of my network. i have done settings as below res_stun_monitor.conf settings: [general] stunaddr = stun.ideasip.com stunrefresh = 30 stun show status Hostname Port Period Retries Status ExternAddr

I am experiencing a "606 not Acceptable" error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. I tried putting stunaddr=stun.ekiga.net into the sip.conf file under [ekiga]. I also tried

Trying to use gtalk: -- Executing [andy at ipkall:2] Dial("SIP/ipkall-00000000", "gtalk/andy-gtalk/+1xxxyyyzzzz at voice.google.com") in new stack [Apr 1 10:41:53] ERROR[2416]: chan_gtalk.c:1934 gtalk_request: No XMPP client to talk to, us (partial JID) : andy-gtalk gtalk.conf [general] context=google-in ; Context to dump call into allowguest=yes stunaddr =

Having some trouble with sip behind a nat. So tried: stunaddr = numb.viagenie.ca in sip.conf. Didn't help so tried stun debug: asterisk*CLI> stun set debug on STUN Debugging Enabled STUN Packet, msg Binding Response (0101), length: 36 Found STUN Attribute Mapped Address (0001), length 8 Ignoring STUN attribute Mapped Address (0001), length 8 Found STUN Attribute Changed Address (0005),

hello, I read http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large anybody could tell me more about this ? Is it available with ARA ? Regards Harry Method 3 Q: If you have your SIP phones registered with SER but your voicemail is handled by asterisk, how do you get the MWI (Message Waiting Indicator) light to function on the phone? A: In sip.conf create a section pointing at your

I'm having the same issue! The difference in my case is Asterisk server has a public IPv4 and the browser is behind a single NAT. I'm forwarding my configuration below (which I posted previously on asterisk-users). How can we debug ICE negotiation? ---------- Forwarded message ---------- From: Vinicius Fontes <vinicius at aittelecom.com.br> Date: 2015-07-27 13:54 GMT-03:00

I have thew same problem. Ui tried with dyn dns in the externip field in sip.conf but I think the Asterisk does not allow this. Unfortunally I have every day a new ip. Maybe I can write a script witch takes my actual ip from externat and put it into the externip field. Maybe this solves the problem. -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com

is there general documentation on how asterisk behaves as a stun client (besides res_stun_monitor.conf) ? e.g.,: * can asterisk use multiple stun servers ? (im interested in availability, not data parity) * what is the relationship between gtalk.conf's stunaddr and res_stun_monitor.conf ? will duplicate queries be sent ? * Does asterisk provide some call (through AMI,

Hi, A brief 6-step guide on how to hardcode a change in the Asterisk source that will allow it to work from behind a nat device. I know it?s messy, but it may prove useful to some people. 1. First punch a whole in your nat device. I just forwarded the port 5060 (for sip) and all ports between 10000 to 10020 (for rtp) to my asterisk gateway. 2. Now make sure your /etc/asterisk/rtp.conf correctly

The Asterisk Development Team has announced the fourth release candidate of Asterisk 1.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc4 is currently scheduled to become the full release of Asterisk 1.8.0. All interested users of Asterisk are encouraged to

The Asterisk Development Team has announced the fourth release candidate of Asterisk 1.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc4 is currently scheduled to become the full release of Asterisk 1.8.0. All interested users of Asterisk are encouraged to

Hi All, I tried to play a little bit with IPv6 to test our VoIP quality software with IPv6 RTP streams. I add "bindaddr=::" to the general section of the sip.conf and netstat shows that Asterisk is listing also on IPv6. My Asterisk server is behind a IPv4 NAT and was working absolutely perfect. But after my bindaddr change I got a problem with external calls. I spend some time to

Em Sat, 18 Mar 2017 16:24:25 +0100, Christian Kivalo escreveu > Am 18. M?rz 2017 18:55:58 MEZ schrieb Nilton Jose Rizzo <rizzo at i805.com.br>: > >Em Sat, 18 Mar 2017 11:36:34 +0100, Christian Kivalo escreveu > >> On 2017-03-18 07:19, Nilton Jose Rizzo wrote: > >> > Em Fri, 17 Mar 2017 22:35:40 -0300, Nilton Jose Rizzo escreveu > >> >> Em Thu, 16

R Users, We would like to announce that Version 1.0.1 of the energy package is now available on CRAN. The energy package introduces a new class of statistical tests based on the concept of Newton's potential energy. Included in the package are * mvnorm.etest (test) and mvnorm.e (statistic) A rotation invariant multivariate goodness-of-fit test, implemented for testing

R Users, We would like to announce that Version 1.0.1 of the energy package is now available on CRAN. The energy package introduces a new class of statistical tests based on the concept of Newton's potential energy. Included in the package are * mvnorm.etest (test) and mvnorm.e (statistic) A rotation invariant multivariate goodness-of-fit test, implemented for testing

Em Sat, 18 Mar 2017 11:36:34 +0100, Christian Kivalo escreveu > On 2017-03-18 07:19, Nilton Jose Rizzo wrote: > > Em Fri, 17 Mar 2017 22:35:40 -0300, Nilton Jose Rizzo escreveu > >> Em Thu, 16 Mar 2017 23:06:08 -0700, Doug Barton escreveu > >> > On 03/17/2017 01:21 AM, Nilton Jose Rizzo wrote: > >> > > > >> > > > >> > >

Am 18. M?rz 2017 18:55:58 MEZ schrieb Nilton Jose Rizzo <rizzo at i805.com.br>: >Em Sat, 18 Mar 2017 11:36:34 +0100, Christian Kivalo escreveu >> On 2017-03-18 07:19, Nilton Jose Rizzo wrote: >> > Em Fri, 17 Mar 2017 22:35:40 -0300, Nilton Jose Rizzo escreveu >> >> Em Thu, 16 Mar 2017 23:06:08 -0700, Doug Barton escreveu >> >> > On 03/17/2017

Hello, If I understand it correctly, when sending INVITE, o= and c= sdp fields are built using p->ourip IP address. At this point RTP packets will be coming to the default asterisk IP address. For the machine with multiple interfaces this could be not the right one (not what we want). Could it be configured (in rtp.conf or in sip.conf per context) ? Thank you. Alex Zarubin --------------

Em Fri, 17 Mar 2017 22:35:40 -0300, Nilton Jose Rizzo escreveu > Em Thu, 16 Mar 2017 23:06:08 -0700, Doug Barton escreveu > > On 03/17/2017 01:21 AM, Nilton Jose Rizzo wrote: > > > > > > > > > Hi all, > > > > > > > > > I already searched for this error on google and nothing > > > > > > I never install

I've attached two SIP debugs in reference to bug #116. They are from today's CVS build. 1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the call. After which, SIP(2) rings for about 30 seconds then stops. 2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before the call is answered. SIP(1&3) are Cisco 7960's and SIP(2) is a Polycom