similar to: Problems with SIP invite due to long ping round trips

I just got setup with vonage.ca with the motorola ata unit.. I fired up ethreal and checked out what's flying over the network... The sniff below would lead me to believe that it might be possible to have asterisk spoof the User-Agent field and register itself? Any thoughts/feedback? Thanks. > > No. Time Source Destination Protocol Info >

Hi all, Please excuse my newbie status I need help in configuring a mediatrix 1204 PSTN gateway with asterisk. Basically each FXO port is configured with a SIP username and automatic transfer extension, which should transfer incoming calls to an asterisk extension. I created extensions corresponding to the FXO port SIP usernames. Port 1 - SIP username - 21383396 - call forward to - 300

Hi, I am faced with this dilema of asterisk not sending an ACK after it receives 200 OK from OpenSER (which is a response to a reinvite request sent by asterisk. Here is my setup Carrier<->OpenSER<->Asterisk1<->Asterisk2 A user is connected with Asterisk1 (through the carrier and OpenSER). On certain dtmf events the call is forwarded to Asterisk2 using the Dial command.

Hello all, I have problems trying to register sipXphone to asterisk. It always print the message : May 26 13:04:33 NOTICE[2781]: chan_sip.c:7691 handle_request: Registration from 'sip:2503@172.25.50.52' failed for '172.25.49.219' Anyone has an idea to help? Asterik seems to fail in register_verify() but i don't know why... My sip.conf is the following : ; ; SIP

My BT100 won't register with my Asterisk server, it always comes back with a 403. I've included my sip_additional (only one to to have the username 2201) and a portion of the sniffer trace (packets 27 & 28). This has me puzzled as I have my SPA-3K working (incoming and outgoing). On my BT100 I get no dial tone, I can't call it (asterisk says the extension is busy) but I can call

I understand Asterisk is more like a B2BUA. But when this INFO request is sent to asterisk, asterisk is supposed to bridge the request to the other endpoint, right? In what situation, it decides to send a reply; in what situation, it decides to bridge the request? What is the role of gateway in SIP world, a proxy, a B2BUA or something else? Thank you, Wei Date: Fri, 18 Mar 2005 12:51:28 -0600

http://bugzilla.mindrot.org/show_bug.cgi?id=158 Summary: ssh-add trips up due to missing key types Product: Portable OpenSSH Version: 3.1p1 Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P4 Component: ssh-add AssignedTo: openssh-unix-dev at mindrot.org ReportedBy:

Hi All, I;m using asterisk 1.4 with FreePBX and a Grandstream 4108. I am noticing a delay calling in and out via the FXO, but calls to local extension are ok. What i noticed when i used ngrep is that, it sends invite but got no response from the server, send another invite but got no response again, then again until it finally gets it. but if you will notice on the 2nd ngrep, the asterisk

http://bugzilla.mindrot.org/show_bug.cgi?id=158 ------- Additional Comments From wknox at mitre.org 2002-03-12 06:43 ------- Created an attachment (id=38) Patch to fix described problem ------- You are receiving this mail because: ------- You are the assignee for the bug, or are watching the assignee.

I'm having trouble connecting a asterisk server to a SIP Express router. Inbound calls to my asterisk server works just fine, but when i try to make outbound calls I get the following error message: Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to WWW-authenticate on INVITE to '"username" <sip:username@mysipprovider>;tag=as5399a078' I'm

We encouter a situation where we need to use SIP info to convey infomation for one end point to another endpoint. I use asterisk to do the test and find asterisk does not forward the SIP info to another endpoint, but act as UAS and returns a 4xx error message. I think asterisk is not right to handle this SIP info message. In RFC 3261 Page 70 "This protocol is designed to be extended.

Hello. I'm trying to use Asterisk in combination with SER, to make the routing proccess to my PSTN-Gateways. I made a simple test defining some extension in my extension.conf, when i made a call my SER (SIP) Server forward the call to Asterisk, this proccess is ok, but when the call is answered i see an INVITE going out from Asterisk to my SER Server, this invite is then passed to my

On 06/14/2017 at 10:17 PM, Joshua Colp wrote: > On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: > > <snip> > >> >> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >> Just one exception - and that's the package in question, which can't be >> seen in tcpdump. >> >> I extended the above patch by adding

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Hi, I tried to cable #21 with a thomson cable modem mta: <-- SIP read from 192.168.153.100:5060: INVITE sip:%2321@195.38.96.5:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.153.100;branch=z9hG4bK1aa77a586 Max-Forwards: 70 Content-Length: 258 To: "#21" <sip:%2321@195.38.96.5:5060> From: sip:15800115@195.38.96.5:5060;tag=da42eb89613306c Call-ID:

Hi. I'm stuck into an odd situation. Here's what happens: 4 Thomson ST2030S 2 Cisco 7912 3 Cisco 7940 2 AAstra 480i Asterisk 1.2.17 Diva 4BRI + chan_capi I've just upgraded yesterday from Asterisk 1.2.13 to 1.2.17. Until yesterday, everything was just fine with 1.2.13. Immediately after the upgrade, *all* the 7940 are no more able to make calls, just receive them, while 7912

Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:01xxxxxx at myip.com I an set an inbound route! It matches a DID number. How can I route an INVITE sip:s at myip.com? The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it

I'm not getting inbound audio from bandwidth.com. Their engineer said the invite that they're sending me looks like this: INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp SIP/2.0. Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>. Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>. Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0. Via:

Hello, I'm receiving some traffic from a Softwitch to Asterisk When I'm hiding the CallerID in the softwitch, everything is all right. When I allow to send the callerid from softwitch to Asterisk (actually, I would like to have it) Asterisk rejects the call with a 407 Proxy Authentication SIP packet. I copy-paste the SIP Invitation: ------------------ Session Initiation Protocol

Hello, I have 2 asterisk servers that are not working well together. One is acting like a registrar (PBX01) for all my PAP2's and other SIP/IAX devices. And the other is acting like my sip gateway (PBX02) to various providers. They are both on a private network and should be trusting each others IP 100%. But the PBX02 challenges PBX01's requests all the time even though