similar to: asterisk-users Digest, Vol 78, Issue 66

Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk stop working after the upgrade. Here is the sip debug: --------------------------------------------------------------------------- <--- SIP read from 208.65.xxx.xxx:5060 ---> INVITE sip:1778xxxxxxx at 10.11.22.77:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport Via:

I am helping a friend on one of his sip trunk and couldn't find the way to resolve his problem. His asterisk's problem is like this: 0. When incoming call to one of his sip trunk, Asterisk reply with "488 Not acceptable here". So the call get dropped. 1. Recently upgraded Elastix with Asterisk 1.4.33 2. Was working fine before the upgrade 3. There are total 4 SIP trunks

Hello, all, I have a Linksys 3102 from a VoIP provider. It use SRV record to register to the provider's SIP server. When I configure this line into my Asterisk, the register doesn't work if I use their domain name. So it like this: If I use register => user:pwd at proxy.provider.com then I got: [2010-05-17 11:47:19] WARNING[2366] chan_sip.c: No such host: proxy.provider.com

Hello. There are so many sound files in /var/lib/asterisk/en. Is there an easy way to let me play back all of them one by one while I am watching CLI to see the current file name? Thanks for help. -- Jian Gao IT Technician SJ Geophysics Ltd. <http://www.sjgeophysics.com> jian.gao at sjgeophysics.com <mailto:jian.gao at sjgeophysics.com> Tel: (604)582-1100

Hi all, I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I compiled it with SRTP support. Everything seems to work OK but I am having a weird issue. I cannot disable SRTP. I tried the /encryption=no/ in /sip.conf /and the /_SIPSRTP_CRYPTO=disable/ on my dailplan and it keeps trying to use the SRTP. Well, right now I have to have/ noload=res_srtp.so/ on my /modules.conf /otherwise

Hello everyone, Recently we've been focusing quite heavily on making Adhearsion[0] faster. To do that, we needed a convenient way to test our Asterisk voice apps. The obvious tool in the Open Source world is SIPp[1]. SIPp is great! Though it's a little clumsy to use sometimes, especially if you're trying to use it to drive interactive calls like an IVR. So to make our own lives

Hello, I have been trying to completely disable music on hold on my asterisk system. When a call is put on hold I do not want any music on hold, but I would like the remote user to get informed of this event (depending on the technology e.g. with a SIP reinvite and an SDP indicating the call is on hold). I have searched and tried out various approaches, but when putting the call on hold

The plastics industry says polycarbonate bottles are safe. http://www.bisphenol-a.org/about/faq.html#g I'm sure Maggie and here friends would say ALL plastic bottles are very dangerous. This lady seems to be at a reasonable middle ground. http://trusted.md/blog/vreni_gurd/2007/03/29/plastic_water_bottles Polycarbonate plastics the kind of bottle you bought contains BPA. "In 2006 Europe

I've been approached to research and develop a system using asterisk. It will be used mainly to provide voice support to about 10,000 IAX clients operating on bank ATMs. So was wondering if there were any financial institutions, banks etc. using * and any comments would be much appreciated. regards joe baptista www.joebaptista.com www.baptista.god

Hi, I have a small problem with Samba in Debian 2.1 (Slink), kernel 2.0.36 and samba 2.0.5. I have a NT Server and a Linux client. I want to get some files from NT Server to the Linux client. They are in the same network and the IP's are: NT Server - 222.222.222.2 Linux Client - 222.222.222.40 Reading samba docs I discover two small utilities:

Hello, did you got your issue solved? I am suffering of the same issue.... On 4/28/07, Hadar Pedhazur <hadar@unorthodox.com> wrote: > > I snipped all of the previous data, as I'm trying to "boil down" > this problem to its essence... > > I turned off the firewall for a few seconds, and still got no > audio. For those that will be suspicious, the commands

Hi, No matter I try, I can not register to Voipwise with Trixbox. It is always in "unregistered" state in sip registry. Here is my last sip trunk configuration: PEER DETAILS: allow=g729 bindport=5060 disallow=alldtmfmode=rfc2833 fromdomain=sip.voipwise.com fromuser=username host=sip.voipwise.com insecure=very maxexpirey=120 pickupgroup=1 port=5060 secret=pass type=peer

Following the setup in Prof.Duncan Murdoch's page, I have successfully compiled the DLL for one Fortran 95 program using Gfortran and got 300 times speed boost. For the second set of fortran programs, However, I have this error message R CMD SHLIB -o jiangang kdtree2.f90 jian.f90 gang.f90 m2c -o jian.o jian.mod make: m2c: Command not found make: *** [jian.o] Error 127 Can anyone

I have a list of CLIDs prefixes that I want to use in a context. Basically, I want to do this but the list of prefix numbers is much longer. List of prefixes (556,557,557,989.....) [custom-inbound] exten => _556,1,answer exten => _556,n,playback(beep) exten => _557,1,answer exten => _557,n,playback(beep) exten => _558,1,answer exten => _558,n,playback(beep) exten =>

Via: SIP/2.0/UDP 10.11.22.161:10000;branch=z9hG4bK-a860600e\x0d\x0a From: Jian Gao <sip:8181234567 at my.provider.com>;tag=7e9c4091bfc704bco0\x0d\x0a To: Jian Gao <sip:8181234567 at my.provider.com>\x0d\x0a Call-ID: daf96244-769f952c at 10.11.22.161\x0d\x0a CSeq: 48998 REGISTER\x0d\x0a Max-Forwards: 70\x0d\x0a Contact: Jian Gao <sip:8181234567 at

Hi, I need to install HP Service Desk with wine, but I can't install MSJVM. Does anyone do this already? Thanks -- Carlos Baptista cbaptista at gmail.com

Good afternoon. I am having trouble trying to set up shared folders in the guest VM using VirtualBox hypervisor. No matter what configuration is defined in my XML, it never works, I mean, no shared folder is added/specifyed, then, inside the guest, when I try to mount the folder, I always get a protocol error. If I use VirtualBoxManage, I can add/specify shared folders easily and mount it with

Hello, I am doing some tests with asterisk on a dual-stack environment. I have some doubts regarding asterisk binding addresses on a server with 2 network cards. According to asterisk documentation: /; With the current situation, you can do one of four things:/ /; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1/ /; b) Listen on a specific IPv6 address.

i.e. iax1-------+ iax2-------| iax3-------|-- * -- MeetMe -- shoutcast radio ....-------| iaxn-------+ would be a great way for corporation to broadcast meetings. board meetings live management meetings live sales demonstrations and all interactive. you get the picture. right now i'm thinking of just patching a cable from the * operator console to a shoutcast server.

Hey all. My boss asked me to implement the following When DID 713xxxxxxx is dialed send an email to mmosier at xxx.com. with the time date and CID included in the email. I know how to code some but am looking for the best way to do this. Sorry I might have asked this a couple months back. I forgot. Mmosier Houston Respectfully Michael D Mosier Ftoc Certified -------------- next part