similar to: Binding to multiple ports in sip.conf

Hello, With asterisk 12 improvements, is it now possible to bind an asterisk SIP stack to several ports ? For instance, to both emit or listen on ports 5060 and 5062. If positive, any hint on how to get more detail about this ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Great ! I'm gonna it try ASAP ! Is there another way (ie not using different ports) to get several trunks to a given ITSP ? Let me explain this a bit further. My setup is: ITSP <---- SIP----> Asterisk <----> Phones For various reasons, I want my Asterisk box to have several trunks/SIP account with my ITSP. First method, is to configure a specific port for each trunk: ITSP will

OpenSSH supports the -b bind_address argument for binding to a local IP address when connecting to a remote host. It's however currently not possible to specify a local port to bind to, something I've found useful at several occasions. Below is an unified diff that introduces the [-B bind_port] option to ssh(1) and a ssh_config(5) style option "BindPort bind_port". This allows

I am using a month old svn version of asterisk 1.2 . I have set bindport=5091 for a sip peer ( type = friend) and nat=yes .. in sip show peer it shows port 5091 for peer but asterisk isnt listening on port 5091 at all . I tried both port=5091 as well as binport=5091 but asterisk does not listen on port 5091 . What am i doing wrong ? -------------- next part -------------- An HTML attachment was

HI All, I am new to asterisk. i want to make outbound calls from asterisk. I tried with many times with the given settings but in vain This is my scenario: I have a *user A* who has registered with sip server(ONDO), I made asterisk to register as a sip client with ONDO, I want to make a call to user A from an extension. My configurations sip.config [general] context=default

Hello all I'm trying to get my asterisk config to listen to multiple ports. This is since some clients have port 5060 blocked by their ISP. Does anyone know how to do this in sip.conf or if it is even supported? Thanks!

Hello, we want to setup the following scenario: - each user has a softphone AND a hardphone - the softphone is started with the operating system - the hardphone is connected all the time using SIP - only ONE extension for each user Both phones should ring when the user is called. We've setup an asterisk 1.4.18 and at the moment only the last registered client rings. In Asterisk 1.2 the

Hi, Is it possible to have multi listening bindport in asterisk? Now days mostly ISPs are Blocking the standard 5060 port so we want to keep option if 5060 is blocked we can ask our customers to use another port. Thank You Abdul __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Trying to configure IAX for use I think I have everything set right. But my IAX phone wont connect. When I run wireshark I'm seeing this Note if above screenshot from wireshark does not show here is a link for it: http://img402.imageshack.us/i/tempe.jpg/ I've tried a variety of setups in my IAX.conf (they all end up with the same issue, tried just bindaddr=0.0.0.0 with

Hi all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060. Also, in the global SIP.conf file bindport=5060 bindaddr=0.0.0.0

I have 4 asterisk servers. They all have local phones on their local network they manage for SIP based conversations. We then have IAX between them all for inter-asterisk connections. This setup has worked well for nearly 2 years now, minor problems here and there but overall very nice. Recently we acquired some Polycom video conference units. I was able to setup our main server to host all

Hi, Andrew. You are trying to solve two tasks: definition through what line the call came and a beautiful display of this information. 1. definition through what line the call came. If the username and password for inbound and outbound registration the same, then try the following: a) delete "register" lines. b) add option "callbackextension=Company1" to Company1 friend

"Jay Milk" wrote ... > >It would appear to me that you could run Asterisk and SER on the >same box. Asterisk's sip-binding can be configured to be 5061 >instead of 5060, and I'd assume SER can be configured as well. Sure, but then the two wouldn't be able to talk SIP to each other. Well, I guess you could use VOCAL, then do this ...

Hi, We have been using Samba File Server (Version 4.3.11 Ubuntu 14.04 LTS) for quite sometime now. We recently installed Samba-AD (Samba AD Version 4.7.6) and made the file server a member of the Domain. Everything was fine till around 11:15 am yesterday. We just added one more share folder and gave access to three users and restarted Samba File Server services - smbd, nmbd and winbindd -

Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but it does work. For prosperity, the SIP service is through Internode. Here is my "extensions.conf" file: exten => s,1,Set(thedid="${SIP_HEADER(TO)}"); ignore this one exten => s,2,Set(pseudodid=${SIP_HEADER(To)}) exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)}) exten =>

Hi Looks like 1.1 version is sensitive to the system architecture/compiler/kernel version. Below is my observation in various linux system I have. As you mentioned, you are not observed the crash in your system, can you please share your system details. And also please comment on the below table/observations. *Machine IP* *optimization flags* *RHEL version* *kernel version* *gcc version*

I am trying to make sure my DID and SIP account details are working properly and engaging the extensions.conf and dial plan. I have a successful SIP session registered: Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922) Asterisk*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060

I just tried decoding with v1.1: ./opus_demo -d 48000 2 opus_encoded_crash.opus out.pcm and I see no issue (including with valgrind). Does the same command-line create problems for you? What compile flags did you use? fixed-point or float, any assembly, ...? Could be assembly here, or even a compiler bug wouldn't be unheard of. Cheers, Jean-Marc On 20/04/15 07:27 AM, Suresh Thiriveedi

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

Still can't reproduce. What OS and compiler version? Jean-Marc On 21/04/15 02:48 AM, Suresh Thiriveedi wrote: > Hi, > > There is no change in the compiler flags. I'm using as it is from the > original code. No change in the Makefile and I believe it is using the > floating point only by default. > > We are using 8k samples and mono so the commands is as follows.