similar to: FW: Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist"

I am not able to get softphone registered (active) with * . new installation , new user Able to get server started , and phone appears to register . gets the SIP reponse 481 message Register SIP '4009' at 192.168.200.10 port 2199 expires 120 Unregistered SIP '4009' Register SIP '4009' at 192.168.200.10 port 9428 expires 120 Saved useragent

I have a PCPHONELINE SIS phone set it up to asterisk Registered SIP '205' at 24.172.221.22 port 2770 expires 120 (Port changes every time) Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 24.172.221.22(24.172.221.22 is my phones IP) Anyone have an idea what the problem is? Jeff

One of our remote user's phone reports frequently: Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from <IP> What can I do ??? bye Ronald

Hi All, I am trying to use iconnecthere to make outbound calls. I am behind a linksys router. I keep getting this error 481 "Call Leg/Transaction Does Not Exist". Does anyone have any prior experience with this problem. Any leads will be much appreciated. Attached are the conf files and logs #SIP.CONF ; SIP Configuration for Asterisk [general] port = 5060 ; Port

Hi, Here is my confing: [out] Exten => _X.,1,Noop() Exten => _X.,2,Dial(SIP/${EXTEN}@peer,60,gcU(do_dtmf_cc-take-call,s,1)) Exten => _X.,3,Playback(tt-monkeys) Exten => _X.,4,Playback(tt-monkeys) Exten => _X.,5,Playback(tt-monkeys) Exten => h,1,Noop(ABCDEFGHIJKLMNOPQRSTUVWXYZ) [do_dtmf_cc-take-call] Exten => s,1,AGI(agi://127.0.0.1:4579/update_call_status?status=60) Exten

Is there a way to specify the outbound leg Call-ID? -- Eric Chamberlain

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such

Hello everybody, I have an odd problem with iptables using a Xen bridge setup. I don''t know if it would be better to post to netfilter Mailing-List. But I hope someone here know how to solve it. If it''s OT here, please let me know. I''ll try to do a little bit ASCII-Graphics to explain the topo better: _________ ________

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name :

Hi, When implementing click2dial feature, I can trigger an Aastra phone to auto-answer using statement like : SIPAddHeader(Alert-Info: info=alert-autoanswer); This is very convenient when trying to reach a distant party (ie through PSTN) The trouble is when 2 Aastra are calling each other over the LAN, this single statement is memorized somehow and both phones (caller and callee) auto-answer.

Dear Sir, I use asterisk 1.8.11 (192.168.100.202)to connect lync server .I use tls port 5068 to connect to this lync server . The tls is ok to establish and I make call from softphone 3200 (register to Asterisk) and dial 9XXXXXXX (9+85225082162) , this prefix will dial to trunk lync_trunk and pass to lync server(192.168.100.14) using tls . But the lync client in opposite side ringing and they

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

Hello friends, I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can access the caller Call ID (fbasename field in voipmonitor cdr) looking at the SIPCALLID variable in asterisk, but how can I access from within asterisk the Call ID of the second leg of the call (the one originating from asterisk to the destination peer)? is there a variable holding this value? Thank you

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I know I've seen this reported already, and I can't remember the fix. I have two ATA186s talking to an asterisk server. When I call in on an outside line, both ring, and I can pick up either and talk. But if I try to call from one of them to the other, the remote end rings just fine in both cases, but then as soon as asterisk bridges the two channels, the remote end sends a

Hello, How can I go to the Leg B channel in Asterisk Dialplan _after_ call ends? I can use Dial G option to go to Leb B channel when call is answered, but how to go here when call ends? Is here any option/function in Dial Plan? Or should I use ast_bridged_channel(chan) to get bridged channel and try to retrieve data I need from internal structures using custom c module and Asterisk API?

Hi all, I have two phones that I've been comparing (different manufacturers). To debug call quality issues on one of them, I've been using calls from the phone to our main DID, so 3 SIP sessions exist (phone <> asterisk then asterisk <> provider, and the provider<>asterisk for the DID). The "bad" phone shows roughly twice the number of packets on the

Hello all, I would like to set a different caller-id for the second leg of a call when doing an originate. For example: Action: Originate Channel: sip/1234 Context: mycontext Exten: 1 Priority: 1 Callerid: "123 <123>" Async: true This sets the caller-id correctly when dialing sip/1234, but I would like to set the caller-id for the second leg of the call (the one that goes to 1 at

Hi We have the following test .call file and test dialplan: I can't set a custom CDR var to a value on one channel leg, and another value on the connected channel leg? Is there a way I can woraround this issue? ## test call file Channel: Local/queue at TiagoGeada CallerID: teste-geada:0:210332450: MaxRetries: 0 RetryTime: 1 WaitTime: 8640 Account: teste-geada Context: TiagoGeada

Hello there, I'm using a Debian box with Asterisk 1.8.13.1 as a DID-PSTN gateway, so I can receive calls in a DID number and redirect it to my mobile line. It has been working flawlessly for a few months, but I have noticed that some calls were not cut after losing one leg (the one with the DID server), and kept the PSTN leg active until I restarted the server (with the unexpected cost