similar to: Asterisk and alternate RTP ports

Hi, I'm trying to configure 2 parallel sip trunks between 2 boxes. However I seem to have the problem that when making a call from Box 2 to Box 1, it sometimes says authentication failed because it is using the username of the other trunk. Here's my configuration: Box 1: [dp-dp2] type=peer username=dp-dp2 secret=mysecret qualify=yes host=box.2.ip.address context=from-internal [e911-dp2]

During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if

How exactly does Asterisk provide E911 service??

One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? ----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part

I just set up a t1 with 2 camma mf 911 trunks on it, and I am having a issue with it. We can call 911 which is routed out these new trunks, and it goes to the 911 center, but they are not getting the ANI and hence "no record found". Our LEC is Embarq, and they say they can see the call come in and send: KP-911-ST and then KP-0-911-ST rather then KP-0-ANI-ST I turned on all the debug

Greetings to all, Can anyone think of a reason that a Softphone would not be compatible with the F.C.C's order for E911? If the user is able to update their address when they move their laptop, etc.

I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten => 911,1,ChanIsAvail(Zap/1) exten => 911,2,Dial(Zap/1/911) exten => 911,3,Hangup() exten => 911,102,ChanIsAvail(Zap/4) exten => 911,103,Dial(Zap/4/911) exten => 911,104,Hangup() exten => 911,203,ChanIsAvail(Zap/5) exten =>

Does anyone have any experience with any SIP or IAX providers that support E911? I'd love to convert entirely to Asterisk at my house, but the lack of emergency dialing has been a major hold-up for me. Thanks in advance for any suggestions! -- Kyle Sexton

Hi, We have been using Voxbone for some time for origination, and they now offer E911 services.? We are trying to set this up and having trouble meeting their authentication requirements. I setup a peer as I normally would, with user/pass as they supplied ("lacoursj", "pass"), but my calls are rejected.? Their support is asking that I follow this auth mechanism: 1st step

Can Asterisks properly handle outbound Enhanced 911? Blake

Hi, We've got a dedicated T1 with two trunks running into our ILECs selective router for 911. Split out of the T1 into two MF CAMA trunks on ILEC side. I'm trying to use asterisks e911 signaling, but I'm having trouble with the dial command. (== Everyone is busy/congested at this time (1:1/0/0)) I'm missing something and I'm thinking it has to do with the hookstate

Hi Folks, I want to setup a follow me routine so that asterisk can call me on the multiple numbers. I tried some of the samples at voip-info but there is a problem with those examples. I dont have coverage in my home area and my cell phone answering machine picks up the phone right away so my home phone never rings. I also want the caller to be able to leave a voicemail and the cell phone

I called bandwidth.com to buy a sip line from them for $30 a month. But they said they will not sell me a sip line since the address on the account is Citrus Heights CA and they can not provide services in that area. On asking further the person clarified that there is no e911 service available in the 916 area code for bandwidth.com But other providers like www.broadvoice.com are able to provide

I *am* doing that, as I assumed it would be required just for the 911 mapping we have provided, but that doesn't change the SIP header. Cheers, j On 05/08/2018 02:41 PM, Khalil Khamlichi wrote: > try setting the callerid with > > same => n,Set(CALLERID(all)=17864089672 <17864089672>) > > ofcourse for each customer you will need to provide his own did. > > >

Right now we I am using Asterisk boxes as a gateway between our Level 3 SIP trunks and our customer PBXs. I love and understand Asterisk but the company I am working for is looking for a more "Commercial" type solution where we can go to a vendor for support etc. I know, we can get Asterisk support etc.. It's not my decision and I sort of get why they are leaning away from Asterisk,

Hi, Am I correct in this.. I want to setup IPTABLES to protect my * box.. The default rtp.conf defines that * will use ports 10000 to 20000.. IAX listens on 5036.. SIP listens on 5060.. I am assuming all ports used by * are UDP.. So I am planning on setting my server to block all inbound traffic except UDP ports 5060, 5036 and 10000-20000.. Am I leaving anything out?? Thanks.. --

I can't find this in the wiki/email history..but I'm sure it's based asked before. The port range define in rtp.conf - is that for connections initiated by asterisk? Or the port range asterisk listens on? Or both? Thanks! MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a client that would like to use asterisk to link their multiples locations together. However, if a person in the remote office dials 911, How can the 911 operator determine WHERE the emergency is?? Since all calss would be going out of the PRI in the main location, the police/fire trucks will show up at our COLO!! I know that there are some that are doing this multi site setup, how did

I would recommend capturing traffic outside your Asterisk server with Wireshark, then running the Telephony/Rtp/Analysize Streams option to determine if you have packet loss at that point in the network. On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote: > Thanks but no Adtran here. > > I do think these stats are indicating an issue, I just don't know how to

I've seen something similar with Adtran SIP gateways. When a re-invite happens the Adtran gets all confused about call stats and marks the pre-reinvite leg of the call as losing large numbers of packets. BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38. Also Adtran SIP gateways appear not to support OPTIONS packets when running in SIP proxy mode, which is