similar to: Anyone help: call leg do not exist err

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,

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

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?

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

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

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

This should be interesting, a double header Friday at 12 Noon EDT, session 2 at 1PM EDT. 1) Pascal Dor?, Media5corp. Pascal will talk about what they've been up to in the year since his last visit. Thanks to the Asterisk mailing list and VoIP community, their Media5fone was able to fix its g722 implementation. I like their product a lot and used it extensively on my old iPod Touch to make and

Hello, I have a question about "directmedia" or "canreinvite", I have experience that whatever I set directmedia=yes or no. After I run sip show settings. all settings looks the same. My question is how I could make sure from "sip show settings" that my "directmedia" configuration is applied. Thanks

I installed pidgin-sipe, since my googling seemed to suggest it would give me connectivity with Lync. I now see the SIMPLE protocol... but not Office Communicator. What am I missing here? CentOS 7, updated. mark

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

Hi all, I would like to know if these troubles affect also other people, for example, I can no longer connect to Sipe (Lync) network by Pidgin since the Glibc has been updated and now I've just discovered that it was no longer possible to take screenshots with gnome-screenshot, I always receive an error message telling: "Unable to save the screenshot to disk: Value for PNG text

Hi, Excuse in advance my poor english. After installing two new servers debian buster with samba 4.9.4 , one as AD ,the second as a fileserver, I was stuck when i tried to connect my users. On a Win10 client , i had the message "windows could not connect to user profile service". The only clue i had was in the event viewer, errors concerning svchost.exe_ProfSvc. After digging

Hello, you can compile opus with "--enable-fixed-point". That saves a cpu ressources. https://github.com/Studio -Link/PKGBUILDs_clean/blob/master/opus/PKGBUILD#L23 -- nice regards, Sebastian Reimers ------------------------------------------ IT-Service Sebastian Reimers Am blanken Boom 14 32369 Rahden Festnetz: 05776-3930000 Fax-Nummer: 05221-17242088 Skype: miete-admin E-Mail:

Hi, in my company we use an asterisk installation with around 50 soft- and hardphones of all kind. From time 2 time the users (almost only Softphone users) report some voice qualities... mostly echoes. These problems do not occur on all PCs at the same time and since setup of our PBX almost any PC user has gotten these issues. When I come there to check, everything is fine again... and I

On 06/11/2015 10:13 AM, Bernard Lheureux wrote: This issue looks to be related to iconv included in the package glibc ! Is that possible to downgrade glibc on a CentOS 6 ? > Hi all, > > I would like to know if these troubles affect also other people, for > example, I can no longer connect to Sipe (Lync) network by Pidgin > since the Glibc has been updated and now I've just

Hi, Here is my confing: [out] Exten => _X.,1,Noop() Exten => _X.,2,Dial(SIP/${EXTEN}@peer,60,gcU(do_dtmf_cc-take-call,s,1)) Exten => _X.,3,Playback(tt-monkeys) Exten => _X.,4,Playback(tt-monkeys) Exten => _X.,5,Playback(tt-monkeys) Exten => h,1,Noop(ABCDEFGHIJKLMNOPQRSTUVWXYZ) [do_dtmf_cc-take-call] Exten => s,1,AGI(agi://127.0.0.1:4579/update_call_status?status=60) Exten

Is there a way to specify the outbound leg Call-ID? -- Eric Chamberlain

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such