similar to: SIP on multiple ports

Hi all, I am trying to get iptables to work for me... I am running asterisk (11.23.0) on a C5 machine. Working fine on port 5060 udp. I have need to tcpenable=yes SIP and run that on port 5068. Since port 5060 is already running I was going to redirect 5068 to 5060. So I thought I could use iptables to do that - but does not seem to be working. 192.168.10.201 is my machine, 192.168.1.3 is the

I have a host 192.168.1.3 that wants to run SIP on 5068 (long story). My host is 192.168.10.201. My host needs to stay on 5060 because of all the other devices I have connected. I tried putting port=5068 in my SIP extension definition but that did not work. So I thought about using iptables to accomplish this: iptables -t nat -A PREROUTING -p tcp --dport 5068 -j REDIRECT

Hi All, In the midst of trying to pilot a deployment of Microsoft Lync (mainly for non-voice collaboration, specifically IM) and integrate it with our Asterisk (11.6.0 if it matters) deployment and a "everything in one place" tool when people are out of the office. I have everything on the voice side playing nice from the Lync side (Lync->Lync, Lync->Asterisk,

Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c:

Hello! Looks like faq, but... Could you , please, point me on how to convert this [cisco] type=friend host=192.168.22.253 insecure=port,invite to pjsip? as you can see another side is very old cisco router, so I can't change anything there. I don't see any examples here

greetings asterisk users :) ive just deployed version 17 and migrated as best I can to pjsip. I can receive calls, and get to my mailbox prompt, however placing calls seems impossible with the following error on dial: Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890) dunkel*CLI> dunkel*CLI> == Setting global variable 'SIPDOMAIN' to

Hey all, Alright. So we decided to not go with Avaya for our next PBX and we are now full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our SIP gateway and call center and Lync is our internal UC and IP-PBX server. I've already got Asterisk tied with our Nortel/Merridian Option 11 with QSig and all is beautiful (except for the Opt11 not receiving names from * but

Hello, I use asterisk to to IAX2 trunking between London POP & Reunion Island pop. I would like to know if it's possible to do a kind of call-limit (i.e. restrict to XX) channels but on a per dialcode and / or destination basis. For example: [trunk] ; reunion proper, i want to send no more than 24 channels exten => _0262XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN}) ; reunion mobile, i want

Hi, I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial command breaks and the call control go to hangup block instead of next priority. The error in CLI says "*Dial requires an argument (technology/resource)*". This error seems legit as there are no contacts for an offline endpoint. The dialplan

I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can't dial. Does Asterisk offer this type of functionality, and if so how? Blake Parker -------------- next part -------------- An HTML attachment was scrubbed... URL:

If understand correctly type=identify is more for sip trunk configuration ? ;[mytrunk] ;type=identify ;endpoint=mytrunk ;match=198.51.100.1 ;match=198.51.100.2 In chan_sip it was just reply 200 OK on keepalive packet without need define trunks. volga629 On Sun, 3 Dec, 2017 at 10:45 AM, Joshua Colp <jcolp at digium.com> wrote: > On Sun, Dec 3, 2017, at 10:42 AM, volga629 at

Hi, It seems that asterisk doesn't send RTCP packets when a call is on hold. Is there any way to get asterisk to send these packets? I'm in the process of setting up a Lync (microsoft voice) server which will use an asterisk box as a gateway. The trunking between asterisk and lync is 'working' however when a call is put on hold asterisk stops sending RTCP packets to lync, and

We have a client that currently has a Microsoft Lync setup. I must admit I know nothing about this setup. What we would like to be able to do is allow the phones on desks connected to this server the ability to dial something that would allow the phone to access an asterisk box to be able to do an agent login over their LAN. Is there any way to do this? Can the Lync server have a SIP trunk to

Hi guys We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk to MS Lync server. If I create the peer in sip.conf the trunk connects with no problem. However, we prefer to use ARA. Whenever we define the peer in our peers table, the trunk does not work, even if we use sip show peer <peer-name> load. Has anyone got any experience of connecting to Lync using ARA?

Has anyone have any experience of running MS Lync 2010 or Office Communicator 2007 using Wine (Ubuntu)?

Are they any gotchas to be aware of in getting Asterisk and Lync 2013 talking to each other using SIP? Or is Lync a pretty standard implementation of SIP? Cheers Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org

I tried to use Microsoft Lync Server 2010 RC with Wine, and it doesn't work. I tried lots of conf edits and restarted the computer multiple times. Are there any alternatives that integrate with Microsoft Office 2010. I am the head of IT for a medium-sized business and I need to come up with a solution. Thank you, barry

Hello, has anyone running a linphone-application on am335x (like beaglebone) with opus codec. My CPU has extreme high load, wenn ist start with opus codec. Is there a possibikity tu optimize f?r this single core ARM? Thanks Helmut Sholz ______________________________ BAYERISCHER RUNDFUNK Rundfunkplatz 1 80335 M?nchen HA IT und Medientechnik Abteilung Systemservice Funkhaus FG Sendungssysteme