similar to: On SIP INVITE answering to IP:port found in Contact: header.

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Surely there must be someone here who can help me with this problem. I have spent weeks trying to get this damned service to work with no luck. I have incoming calls working, but no outgoing. If get outgoing working then incoming don't work. I have sent this problem to this list a couple of times with little or no response, and I _really_ need some help to sort it out. I have an asterisk

I've been struggling with a Trixbox running Asterisk 1.6 for one of our customers as of late. The service provider in question is using BroadWorks and requires a single trunk registration for the trunk group. We have 4 users(lines/numbers) in the TG. The sip trunk is setup as follows: type=peer host=192.168.1.1 fromuser=<tgid> fromdomain=<sip domain> dtmfmode=rfc2833

I have a SIP account that I can successfully register with XTEN and a Sipura-2000. I have yet to be able to get it to authorize with *. My XTEN looks like: Username: 001234 Password: xxxx Authorization Username: 001234 Domain: domain.net Register with domain:

Hello, Have a weird problem when using asterisk (1.0.6). There are certain numbers I cannot dial when using asterisk with my broadvoice account. No problems with inbound. With outbound calls, I can call some numbers (for example broadvoice customer support number) and unsuccessfully with some. However, when I configure my account directly on x-lite, I dont see these outbound problems. Here is a

I am looking for a sample sip configuration of a SIP provider that runs Broadsoft VoIP switch. My sip provider is Conecta from Brasil, that only give me a SIP IP address to configure my asterisk box, when I call them for support or authentication data to load on my sip.conf, they say me that I don?t need such data, so, anyone knows how I would configure my Asterisk box against a Broadsoft peer?

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

Thank you Joshua. I tried setting the from_domain for the endpoint, but it still sends the internal ip address for the INVITE's From field [acl1] type = acl deny = 0.0.0.0/0.0.0.0 permit = variousaddress permit = bluipaddress [transport1] type = transport bind = 0.0.0.0 protocol = udp [BLUIPIN] type = aor remove_existing = yes contact = sip:bluipaddress [auth7] type = auth username =

??? I am having a really bad day trying to get incoming calls to work on Asterisk 13 with PJSIP.? We just migrated from Asterisk 1.8 where everything was working but there seems that something got lost in translation.? No matter what I try I always get a 401 Unauthorized message when receiving a call from the PSTN provider.? I can make calls and the registration is working.? I have tried to

I am trying to configure a connection to BluIP. I am able to make incoming calls work. However outgoing calls are not working. For the Outbound Registration, I noticed the contact field is always the internal IP address of my pc instead of mycompany dot com I can Originate (using AMI) to my Vitelity trunk (IP based authentication). However, when I Originate to my BluIP, it is being rejected.

Hello everyone, Well here is my initial posting to the list, and I will admit Asterisk is new to me. I just got everything running here a couple days ago, so still learning the ropes for sure. OK, here is my problem. Currently I have it setup talking to a couple Cisco IP phones, and some Xten softphones, this works great. I also got an account with FreeWorld Dialup using IAX2 and that

Hey Folks, Is it possible to transfer an incoming call back out without a "trombone" effect. For instance; Caller dials my broadvoice # --> Asterisk Answers and plays a menu --> the caller selects an option --> asterisk transfers the call to my cell phone via broadvoice and removes itself from the equation so I end up with... Caller --> Broadvoice --> Cell Phone Vs.

Like everyone else who used asterisk with broadvoice, my outgoing calls died last week. I made the appropriate changes, and now basic incoming and outgoing calls are working. However, I have a few call-forwarding rules that are no longer working. It's certainly no coincidence. I can dial to all these number directly, but the problem only appears when there is an incoming broadvoice call, and I

Hey All, I'm just daydreaming here.. but what's the status of SIP-T in Asterisk? I haven't been able to find a whole lot of info on SIP-T but seems like just an extension of SIP. Right? Now if I had a PSTN Gateway (that is a SS7 gateway) that supported SIP-T, could I signal * with SIP-T from it and have asterisk utilize MGCP to sieze a particular DS0 on a remote DS1? Hmm.. What am

Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in

I have been going crazy with this also since Sat. Our server was working perfectly with BV but will now not place calls to BV. Incoming from BV works fine. I felt sad rebooting it today, it had been running for almost 200 days! Here is my error message from the console... Notice I am running today's CVS Connected to Asterisk CVS-03/08/05-14:32:39 currently running on com (pid = 1624)

It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that?

Hi, I am currently making the transition from gcc 4.2 to clang for the projects (mostly C++) I am working on. I think I discovered a major optimization bug in Clang++. I managed to create a simple (60 lines of code) test case which exhibits the issue. When you compile this file for i386 with optimizations turned on (O2, O3 or Os), you get an unexpected result. When you compile it for x86_64, or

Hi, experts for example, the below case: You can see a lot of back and forth MTU probe packets been exchanged between tinc nodes, but it’s weird that, from the debug log, one line shows "No response to MTU probes from node1”, but it indeed received a lot of MTU probe response, and finally it get the conclusion of "Packet for node1 (1.1.1.1 port 443) larger than minimum MTU”.

https://bugzilla.netfilter.org/show_bug.cgi?id=1410 Bug ID: 1410 Summary: STATELESS, rules with notrack into a map Product: nftables Version: unspecified Hardware: x86_64 OS: Debian GNU/Linux Status: NEW Severity: enhancement Priority: P5 Component: nft Assignee: pablo at