similar to: T.38 Problem

Dear All, I'm trying to send FAX to an endpoint Behind NAT...The scenario i the following: PSTN_GW-->Asterisk-->asterisk-->OpenSIPS-->Endpoint behind NAT.. The FAX is failed and I got the following error log on asterisk: Mar 20 09:21:09] WARNING[20388]: chan_sip.c:12409 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog

Hello all, I've been using * for quite some time and yesterday I decided to add sipbroker to my config. It was pretty simple and it works for some numbers (e.g. I can call *258-9123, UK date & time - which is on the "phone numbers you can call" page -) but fails for some others. For example I've got a friend who's at freephonie so to call him, I would dial

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack --

Hi List, Ive got a few * boxes connecting together, one box is doing AgentCallBackLogin() and then the 2nd box is holding some phones at a remote site. I have users login to the main box and * shows the user is logged into a extension that resides on the other box, problem is, when I go to make a call to a agent, I get "May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite:

Hi. I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and for some reason it's simply not doing it. I've even resorted to reading the source code to try and work out what I'm doing wrong... In channels/chan_sip.c I find: * SIP Dial string syntax: * SIP/devicename * or SIP/username at domain (SIP uri) * or

Hello, I need help for that error message: ?chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to? My network is: Client1-- -----------asterisk1------asterisk2 Client2-- ? With client1, I do a call ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Asterisk1 forward the call to

Greetings, i've been posted a message to this list in july, which had one response. Thanks for that idea! Unfortunately asterisk is only a hobby, and did not have much time dealing with the problem since. My original letter was long, i wouldn't post it again, the archive url is http://archives.free.net.ph/message/20070710.053008.c02209c0.en.html Since than i've upgraded to

I have 2 asterisk boxes connected via SIP box 1 sip peer connected to box 2 (ip addresses intentionally removed) [ast20] type=friend host=x.x.x.20 insecure=very context=subscriber dtmfmode=inband qualify=no canreinvite=no disallow=all allow=ulaw box 2 sip peer connected to box 1 [sbb19] type=friend host=64.1.8.19 insecure=very context=inbound dtmfmode=inband

Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615 at sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1' --

In an attempt to connect our Asterisk 1.6 phone system with another phone system called "Broadsmart", they gave me credentials to register to. Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365) watermelon*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time {broadsmart_ip}:5060 N

-uzzi PS: If you're not seeing any connection information, be sure to double-check the IP address is correct. Learned that lesson the hard way =\ On Sun, Jan 31, 2010 at 5:51 PM, Jim Rosenberg <jr at amanue.com> wrote: > Let's say I have two Asterisk boxes, A and B. I am trying to get A to do > SIP registration on B, so an extension for A can dial SIP phones covered by >

On Mon, Dec 15, 2014 at 3:34 AM, Recursive <lists at binarus.de> wrote: > <snip> >> For asterisk 1.6 & 1.8 you would need to set 'canreinvite=no', I don't know what Asterisk 13 will do with this setting. >> > I suspect Asterisk 13 will just ignore it. To make things worse, there seems to be a configuration directive named reinvite (not a typo); I

On Sunday 25 October 2020 at 16:27:00, Antony Stone wrote: > Hi. > > I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and > for some reason it's simply not doing it. I've made a bit of progress - I can now get it to authenticate, although it's still not dialling on to the correct number. > I've even resorted to reading the source code

In a 1.2 release of asterisk, I've had no problem connecting to a Broadvoice SIP peer, to allow routing outgoing calls from Asterisk to Broadvoice. Now, with the same SIP configuration, I cannot establish the peer. I've enclosed a SIP log in the hope that someone can help me analyze this failure. I'd guess the issue is NAT related and wondering if someone can spot a problem in the

I have an updated asterisk 1.8 server running on Freebsd 8.1, and connecting through a Freebsd 8.1 pf firewall with a dumb modem adsl connection (in other words FreeBSD is doing all the hard work). I am trying to connect with Internode nodephone, but they aren't really willing to spend the time to work it out (depending on who you get to talk to), and they reckon its all working as it

Asterisk 1.4.32 (Also 1.4.26, 1.4.33) Broadvox ITSP (xxx.xxx.xxx.xxx) Linksys 2102 (yyy.yyy.yyy.yyy) Both peers : canreinvite=yes t38pt_udptl = yes I'm having some trouble getting a T.38 fax call established with Broadvox. During negotiation, Asterisk sends a SIP re-invite (T38 switchover) to Broadvox with the Asterisk server's IP address in the Connection Information (c) instead of

Hi guys, sorry for the long e-mail, i'm only trying to give as much information as i think is relevant to my problem (console log, sip.conf and extension.conf parts). i've been practicing with callback for a while, but i'm at a dead end. I hope somebody can help me to move on. i have troubles getting two calls bridged together. Scenario is the following: - asterisk calls my

Calling my home numbers has always worked. Till now. The Asterisk CLI show the following : [Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to '<sip:092xx90xx at 85.xx.xx.xx>;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of

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

voipstunt: "Forbidden - wrong password on authentication for INVITE to ...." I have paid, the password was not changed, ... I have no idea why. Is there anything what I can do to get this "failed" call over to another provider, so that the user can complete the call? (Dialstatus was an idea, but the line does not show up in CLI) [Apr 5 09:22:36] -- Executing