similar to: Asterisk Re-invites - how to tell ?

Hi all, Is it possible to set the maximum number of simultaneous calls on an IAX trunk? I want to allow a maximum of 4 calls on our IAX trunk to a remote office, and then route any additional calls over the PSTN after that. I was thinking of keeping track of a count using the AstDB, but I'm not sure how this would go if I had 4 calls incoming on the IAX trunk... Anyone implemented

Hi all, Is there anyway you get Asterisk to let you park a call a second time via pressing the # key + 700 (or whatever is in features.conf) ? If you get a call out of the parking lot & hit # again it just sends the DTMF tone down the line and nothing happens.. A few posts have hinted that you cannot blind transfer a parked call, but surely there must be some way around this as I can

Trading up to 10 google wave invites for a fast, powerful Rails host that provides me with a top level domain: ie: webwebwebwebweb.com or wwwww.com Or similar offers. Send me an email or post here. -- Posted via http://www.ruby-forum.com/.

Hi everybody, I wanted to remind you to a) To request invites to the Xen Hackathon of you have not done so yet b) If you have an invite sign up at http://www.regonline.com/Register/Checkin.aspx?EventID=1211624 (I noticed that at least 5 people who have been granted an invite have not actually signed up) c) Note that about half of the available invites have gone d) Please also add

Endpoint sends an INVITE Asterisk send an INVITE to the Carrier Carrier is down, does not even sends ACK PJSIP sends several INVITES End point sends <--- Received SIP request (397 bytes) from UDP XXXX::50187 ---> CANCEL sip:xxxxxxx at xxxxxxx SIP/2.0 Via: SIP/2.0/UDP xxxxxxx :50187;branch=z9hG4bK-524287-1---fbad0437cf02653d;rport Max-Forwards: 70 To: <sip:xxxxx at xxxxx> From:

Hi all. I have an issue with T.38 and re-invites. Topology: provider -> A (asterisk 1.6) -> B (asterisk 1.6) -> extension -> -> (software fax, gateway whatever). When between A and B trunk is canreinvite=no everything is working smooth. When I switch canreinvite to yes, it stop working. Do you have any idea where the issue can be? Any help will be much appreciated. Marek Soha

Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have

I am using asterisk 1.4.18 (server A ) and have it store records in mysql database . One of my client uses predictive dialer ( asterisk 1.2.26 based and server B ) which makes many calls . B registers with A over sip and there is no nat involved If i re-invite rtp from server B to my carrier ( server A in between ) I saw many calls having duration of 0,1 or 2 seconds on server A's cdr but

Hi all, As i have already posted, i'm noticing a strange problem on my * server: It seems to me Asterisk is simply ignoring some of invites sent from my xlite 3.0. If i dial 2XXX numbers, all ok. If i dial 4XXX numbers that aren't accounts on asterisk i get answer from asterisk. If i dial 4XXX numbers that exist on my server nothing happens and i get call failed: Request timeout.

I have a SIP trunk to Broadvoice. My Asterisk box (1.4.13) is on public addresses behind a firewall. Originally it was behind a Linksys WRT54G running sveasoft. Sveasoft really can't NOT do NAT even when you turn it off. My Asterisk box is defined as the DMZ box to Sveasoft and it seemed like it was leaving all packets alone. Now I switch to a Centos-based firewall configured with

I'm getting invites to very high ports every 30 seconds from a particular ip address: Retransmitting #10 (NAT) to 5.199.133.128:52734: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734 From: <sip:37120116780191250 at 67.80.191.250>;tag=1872048972 To: <sip:3712011972592181418 at 67.80.191.250>;tag=as3a52e748

On 12/30/2017 08:18 PM, Dovid Bender wrote: > Script kiddies trying to find vulnerable systems that they can make > calls on. Lock down the box with iptables and use fail2ban to block > them. The via is probably bogus unless a box at the DoD was comprimised. > > > > On Sat, Dec 30, 2017 at 6:49 PM, sean darcy <seandarcy2 at gmail.com > <mailto:seandarcy2 at

I currently have OpenSIPS set up with users and most of my call handling. OpenSIPS won't be able to handle things like Call Park, Hunt Groups, ACD, etc. So I want to send these types of requests to Asterisk. I also want to set Asterisk up as Multi Tenant. So my question is How can I send requests to Asterisk and have them funnel into the specific context for that specific Tenant? So if

Hi openssh-unix-dev at mindrot.org! Your friend om from , just invited you to his online photo albums and journals at Zorpia.com. So what is Zorpia? It is an online community that allows you to upload unlimited amount of photos, write journals and make friends. We also have a variety of skins in store for you so that you can customize your homepage freely. Join now for free! Please click the

When a call is place to xxx9931211 from the pstn the call proceeds normally until asterisk issues the Second INVITE, which is MESSAGE 14, and instead of call being sent with INVITE sip:xxx9931211@proxy.yyyyy.net SIP/2.0. It gets sent with INVITE sip:xxx9931211@yyy.33.165.201:5060 SIP/2.0 and this seems to cause SNOM proxy to return the packet without a Record-Route entry and then asterisk starts

Many of my dialplan scenarios involve transferring incoming calls back out to other numbers. For reasons of call quality and bandwidth, I would like for the calls to be reinvite'd to bypass my server with the audio channel. What I am seeing is that my server does indeed send the reinvites, and I get OK responses, but the audio never stops passing through my server. I've been fooling

Hi All I'm getting tens of thousands of these messages ever hour in the Asterisk CLI for Asterisk 13.22.0: [Apr 20 15:59:46] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on 1924200000-502043860-301870737 on non-critical invite transaction. [Apr 20 15:59:46] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on 301794058-652332923-1834701069 on non-critical invite transaction. [Apr

Bedankt voor uw bericht. Online4You is sinds 1 augustus niet meer operationeel. Per e-mail hebben wij u geinformeerd over de omstandigheden en uw opties. Helaas kunnen wij u niet meer helpen, uw mail wordt niet doorgestuurd en/of beantwoord. Indien uw abonnement is overgenomen door KovoKs, kijk dan voor contactgegevens op https://www.kovoks.nl/. Dank voor uw vertrouwen de afgelopen jaren! Met

15 February 2013 Hi, Rudi Ahlers is inviting you to join Freelancer.com (http://www.freelancer.com/users/7032909.html?utm_campaign=new_freelancer_invite&utm_medium=email&utm_source=freelancer&utm_content=new_freelancer_invite) Rudi Ahlers

Hi speex-dev! Your friend loverays from , just invited you to his/her online photo albums and journals at Zorpia.com. So what is Zorpia? It is an online community that allows you to upload unlimited amount of photos, write journals and make friends. We also have a variety of skins in store for you so that you can customize your homepage freely. Join now for free! Please click the following