Incompatible destination sip. All causes exposed here are defined in JsSIP.

Incompatible destination sip 21: 603: CALL_REJECTED: INCOMPATIBLE_DESTINATION: incompatible destination [Q. What occurs is th You are right. > I You signed in with another tab or window. 38 is a NO GO REJECT and it probably says something to that in the debug logs. si. com , I'm using JsSIP from a webpage to make a SIP call to FS, using OverSIP as a Websocket->SIP proxy. 15. 2 client A to B: A: Google Chrome broswer built with web: SIP. 11. NON_EXISTENT_CUG. Via: SIP/2. 127. there should also be something there about disable transcoding, make sure to turn that off. C. 0 Via: SIP/2. incompatible destination. 604732 98. 88 – Incompatible destination This cause indicates that the equipment sending this cause has received a request to establish a call which has low layer compatibility, high layer compatibility, or other compatibility attributes (e. server: Freeswitch 1. info xmpp: jungle-boogie at jit. Mandatory information element is missing AST_CAUSE_MESSAGE_TYPE_NONEXIST: 97. JsSIP provides a set of causes in order to make the user aware of what made the request or session fail. c:4035 Originate Hello There, I'm trying to implement JSSIP with Telnyx SIP endpoint. Building a community of users to advance their knowledge and understanding of voip through sharing, learning and supporting each other. 0/TCP 217. SIP Status Code. c:3436 Originate Failed. 2:49881 INVITE sip:auto_to_user at 87. 503 Service unavailable. Information element /parameter 27 502 DESTINATION_OUT_ OF_ORDER destination out of order [Q. > That means compact headers, eliminating extra headers, or just don’t use > such a big SDP by eliminating CODECs you don’t need. {IP ADDRESS}:38532 [CS_EXCHANGE_MEDIA] [INCOMPATIBLE_DESTINATION] > freeswitch About us. The term "not functioning correctly" indicates that a signal message was unable to be delivered to the remote party; e. 142 answer() ----- It looks freeswitch send with ICE to webRTC. Forbidden 21 – Call rejected. Previous message: [Freeswitch-users] INCOMPATIBLE_DESTINATION? Next message: [Freeswitch-users] Javascript self. SIP Status Code to ISDN Cause Code Mapping. 729a and G. Slandered SIP doesnt support m line in a format like UDP/TLS/RTP/SAVPF. The DISCONNECT message cause code can vary. Thats why you're getting that the phone defaults to sending > crypto in AVP which is Also: When call is hang up that involves a SIP channel, Asterisk sends the extra SIP headers “ X-Asterisk-HangupCause ” and “ X-Asterisk-HangupCauseCode ” in in Sofia SIP will normally raise USER_NOT_REGISTERED in such situations. sip SIP Equiv. SIP Request Failure Response Codes to ISUP Q. Cause: INCOMPATIBLE_DESTINATION ===== I'm not that much good with freeswitch, can anyone help me or suggest something? Regards, Arysh. Incompatible destination. JS I receive 488 Incomartible destination. to mapping Q. 850] This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. We strip off the UDP/TLS/ part from the m line. [Freeswitch-users] 488 Not Acceptable Here (INCOMPATIBLE_DESTINATION) with JsSIP and OverSIP + FreeSWITCH Paan Singh paan. Now there are two problems. I'm able to Register successfully, but when I make a call from JsSIP UA to FreeSWITCH, I get a 79. 90. causes namespace and hence, any cause received in an event providing a cause field can be compared against it. Bad Request 41 – Temporary failure. I was able to fix the certificates and stop the codec error, however it's still getting INCOMPATIBLE DESTINATION and hanging up as a result. com> wrote: > We normally use "proxy_media When you restart/fsctl crash freeswitch the CDR that's written says INCOMPATIBLE_DESTINATION. 1 -1 SIPTrunk Endpoint(f196b918) received CMReleaseComp > > A call to a SIP registered endpoint (VoIP handset) randomly fails, it rings, then when it is answered the call fails. Anthony Minessale 2010-01-21 18:53:23 UTC. Service unavailable. 130. de Tue Oct 26 08:51:04 PDT 2010. Enumeration Cause Description Try to add this line in the vars. This is for webRTC. 1. s=sip call c=IN IP4 213. 7, Incoming sip calls failing with Incompatible Destination, but ICR is setup Thread starter DaveElder; Start date Aug 14, 2016; Status Not open for further replies. Description. 850] This cause indicates that the equipment sending this cause has received a request to establish a call which has low layer compatibility, high layer compatibility or other compatibility (+) If the cause location is ‘user’ than the 6xx code could be given rather than the 4xx code. Incompatible destination 503 Service unavailable N/A 102 Recovery on timer expiry 504 Server time-out N/A SIP Status. put the ip address for our call manager in the Domain field. The freeswitch. Response received Cause value in the REL. 'sofia profile <profilename> siptrace on' from the CLI, replace on with off Q. high layer compatibility or other compatibility attributes (e. 503. Nobody likes a guessing game on a technical forum. [Freeswitch-users] Error: 88 Incompatible Destination after upgrade A E G 2012-08-03 04:01:30 UTC. --Yossi Neiman Case 1: SIP trunk between IP office and Call Manager. The cause is: "Cause 88, Incompatible destination". The new version code list that follows Q. E>;tag=rccD75cXD61Kr > To: <sip:0771221122 at voipprovider. x On Friday, November 6, 2020, Maciej Bylica <mbgatherer at gmail. Turn on late negotiation. Previous message: [Freeswitch-users] vvx 410: INCOMPATIBLE_DESTINATION Next message: [Freeswitch-users] vvx 410: INCOMPATIBLE_DESTINATION [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging LinkSys3102 and X-Lite Anthony Minessale anthony. Incompatible destination AST_CAUSE_INVALID_MSG_UNSPECIFIED: 95. When you look at the SDP for the codecs requested by the remote and compare them to what FreeSWITCH is offering and then you can see what it is that you need to activate. The SBC maps SIP For calls that require SIP-H. [INFO] mod_dptools. It would be better if you could give at least SIP traces for the call. Message type non-existent or not implemented. 850;cause=88;text="INCOMPATIBLE_DESTINATION" you can try to flush the "sip-auth-fail" and/or "sip-auth-ip" chains with these commands iptables -F sip-auth-fail and/or iptables -F sip-auth-ip. The issue was in m line. Then in x-lite I created a sip account and put in my username and password. We confirm this SIP message with "ACK" and then the IP Office reports on the "CMLine Rx: v=0" a "CMReleaseComp". 38 stuff in it, I bet its getting tripped up because T. the cause value received in the H. If there is a discrepancy in the IP address/port provided and FreeSWITCH does its best to avoid transcoding. 323 call leg. Here is SIP. [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging LinkSys3102 and X-Lite Juan Antonio Ibañez Santorum juanito1982 at gmail. Invalid message, unspecified. de SIP/2. 80 s=FreeSWITCH c=IN IP4 192. When there is an incoming call from CUCM to IP Office, IP Office displays 503 service unavailable, destination incompatible cause 88. Unauthorized 21 – Call rejected (*) 402. A 488 means "Not Acceptable Here", it usually indicates that there was no codec intersection. Calls do not appear to be reaching 3CX. 729b are incorrect and your device is at fault only other phones of the same type would > > Freeswitch get's their SDP on the bleg (their software have modified the > trace a bit): > > SIP/2. Protocol Discriminator: Q. se [Freeswitch-users] 488 Not Acceptable Here (INCOMPATIBLE_DESTINATION) with JsSIP and OverSIP + FreeSWITCH I'm using JsSIP from a webpage to make a SIP call to FS, using OverSIP as a Websocket->SIP proxy. X-Lite Phone is ringing and after pickup sometimes I get an error All groups and messages Failure and End Causes. 0/UDP 212. unread, Jul 23 I think you need to set codecs in inbound-codec-prefs and outbound-codec-prefs field in sip profile module Now I am trying to send fax using span_dsp modules, my issue is when i receive a fax perfectly but in fs_cli i receive 'INCOMPATIBLE_DESTINATION' ,i need to 'NORMAL_CLEARING'. The values recorded in RADIUS Stop records for the disconnect cause Incompatible Destination 88 Invalid Revision Interworking Unspecified 111 No Permission SIP-SIP Calls. I'm able to Register successfully, but when I make a call from JsSIP UA to FreeSWITCH, I get a 180 Ringing, but after that I get a 488 Not [Freeswitch-users] 488 Not Acceptable Here (INCOMPATIBLE_DESTINATION) with JsSIP and OverSIP + FreeSWITCH Paan Singh paan. It indicates that the switch/PBX clears the call immediately after the CALL PROCEEDING message. 6. 26. Invalid message, unspecified; 96. com Tue Aug 6 17:41:30 MSD 2013. I'll try if Anthony really needs them. SIP/2. 850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0. Find out why avaya send Incompatible Destination. Samir Doshi. [INCOMPATIBLE_DESTINATION]" v=0 o=FreeSWITCH 1343934933 1343934934 IN IP4 192. 142 answer() Cause: INCOMPATIBLE_DESTINATION freeswitch at iZ23lkvsnwpZ> EXECUTE sofia/internal/1004 at 120. , Previous message: [Freeswitch-users] Call from SIP server to external SIP provider: 488 INCOMPATIBLE_DESTINATION Next message: [Freeswitch-users] double 401 Messages sorted by: It can be suitable, but every SIP server is configured with a list of the codecs it accepts/offers so that list can be different depending on which server you're If a Reason header as described in IETF RFC 6432 is included in a SIP 4xx, 5xx, 6xx response, and you have enabled the sip-config, sip-response-code parameter, the SBC maps the Cause Value of the Reason header to the ISUP Cause Value field in the REL message, as follows: [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging LinkSys3102 and X-Lite Achim Stamm stamm at lyth. 95 INVALID_MSG_UNSPECIFIED Saved searches Use saved searches to filter your results more quickly 文章浏览阅读1. 0. You switched accounts on another tab or window. Previous message: [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging LinkSys3102 and X-Lite Next message: [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging LinkSys3102 and X-Lite The call flows are: >> >> the problem is that webRTC calls sip phone without any problem, but sip >> phone calls to webRTC than failed. The term "not functioningcorrectly"indicatesthat a signal message was unable to be delivered to the remote party; e. I cant figure out what im doing wrong. When I set my ITSP to only offer PCMU everything works as expected. 1>;tag=4f3773fe962fef8f Content-Length: 0 11:00:17 2679435mS SIP Rx: TCP 192. The Yealink extension (201) has this dialstring assigned following guidance from documentation (tweaked) (=) ANSI procedure ISUP Cause Value SIP Response Normal event 1 – unallocated number 404 Not Found 2 – no route to network 404 Not Found 3 – no route to destination 404 Not Found 16 – normal call clearing --- (*) 17 – user . 400. Then associated the Sip device to my end user. Invalid transit network selection. 54. To: <sip: 32091846@10. Destination Pattern=918221[67]. Now I am trying to send fax using span_dsp modules, my issue is when i receive a fax perfectly but in fs_cli i receive 'INCOMPATIBLE_DESTINATION' ,i need to 'NORMAL_CLEARING'. 258755 217. 401. 234. , the user may wish to The customer dials, Invite goes out, then a little SIP Ping Pong (trying, ringing, etc) and finally Swisscom confirms with "SIP OK" the actual connection setup. com> wrote: > Now we have some routers and i got INCOMPATIBLE_DESTINATION from AS5350 > cisco router after routeing call to the The call flows are: > > the problem is that webRTC calls sip phone without any problem, but sip > phone calls to webRTC than failed. 1 7443 behind apache ws_tunnel. See ISDN Cause Codes or ISDN Switch Types, Codes, and Values; For a translation table from ISDN codes to SIP, see RFC 3398; Version notes. Previous message: [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging LinkSys3102 and X-Lite Next message: [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging It doesn't happen every time and it's on a production system with a bit of volume. 850 Reason with cause mapping: sip-status —Set the SIP response code that you want to map to a particular Q. I found the solution, was missing "Rerouting Calling Search Space" in device configuration. minessale at gmail. xml <action application="set" data="media_mix_inbound_outbound_codecs=true"/> Regards, Sonny > On Nov 22, 2020, at 4:03 AM, Maciej Next message: [Freeswitch-users] incompatible destination Messages sorted by: Hello, The extension 7006 was working few days ago and now it not working,rest all the phone are working properly. 190;rport;branch Acc. , data rate) which cannot be accommodated. 237252 [NOTICE] switch_core_state_machine. 850;cause=88;text="INCOMPATIBLE_DESTINATION" Anyone seen this before? Situation is, client calls in, phones ring, someone attempts to answer, call gets hung up on, and phones PRI Hangup Codes. Here's what I've tried: Enforcing a hardcoded value for SIPTAG_TO: nua_bye(op->op_handle, SIPTAG_TO(sip_to_make the call with sip trace on? Perhaps the incompatible destination comes from an endpoint. 90* *variable_plivo_request_uuid: 40f1a4ba-d113-11e2-a534-000c29498cfc* *variable_plivo_answer_url: Hi ##### This is the inbound part to freeswitch when they are calling me: 2019/02/12 12:44:00. 1. 931 (8) len=36 < Ext: 1 Cause: Incompatible destination (88), class. Common cause codes include Invalid information element contents and Incompatible destination. 168. Yes, G. therefore a bit hard to get SIP traces. Enumeration incompatible destination [Q. FS do say this: Perhaps the incompatible destination comes from > an endpoint. !--- Outgoing Q. 127 – Interworking, unspecified. Post by George K. Payment required 21 – Call rejected. Non-Existent CUG. Cause 88 Incompatible destination - This cause indicates that the equipment sending this cause AST_CAUSE_INCOMPATIBLE_DESTINATION: 88. SIP Response. 67:5060;branch=XYXYXYXYXYXYXYX Record-Route: <sip:reg. Go into your internal sip profile and make sure that codec negotiation is set to generous. session. INCOMPATIBLE_DESTINATION. 850 to SIP Responses based on RFC3398: Cause 41 Temporary failure - indicates that the network is not functioning correctly and that the condition is not likely to last a long period of time, e. Invalid message unspecified AST_CAUSE_MANDATORY_IE_MISSING: 96. com [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] You signed in with another tab or window. agiv cell-buddy ! com> Date: 2014-06-05 15:06:31 Message-ID: CANba_avU15dCGg-9jDQRG7oauqhDUY8A9r3r=RMK7Sk5cndNcg mail ! gmail ! com [Download RAW message [Freeswitch-users] INCOMPATIBLE_DESTINATION? Michael Jerris mike at jerris. 10. Aug 14, 2016 Cause=88, Incompatible destination 13:57:51 1522413mS Sip: c0a8010a0000041d 18. This is a follow up to the previous problem. server. js通过浏览器呼叫浏览器侧无生意 其次，根据SIP信令状态码，推测原因有二，首先是编码不兼容协商失败，其次是地址问题，因为WebRTC使用STUN处理防火墙穿透和NAT地址穿 2. Reason: Q. You signed out in another tab or window. U 192. > > 'sofia profile <profilename> siptrace on' from the CLI, 2013-07-21 16:07:53. getVariable Messages sorted by: If a Reason header as described in IETF RFC 6432 is included in a SIP 4xx, 5xx, 6xx response, and you have enabled the sip-config, sip-response-code parameter, the SBC maps the Cause Value of the Reason header to the ISUP Cause Value field in the REL message, as follows: 88 – incompatible destination. Sayyed Mohammad Emami Razavi < emamirazavi at gmail. data rate) which cannot be accommodated. W3C CSS3 CSS3 These are typically directed to the google group. 971775 > Cause: INCOMPATIBLE_DESTINATION > freeswitch at iZ23lkvsnwpZ> > EXECUTE sofia/internal/1004 at 120. 90. 1053. The values recorded in RADIUS Stop records for the disconnect cause SIP Request Failure Response Codes to ISUP Q. de Fri Oct 29 03:53:28 PDT 2010. ISDN Release Reason. min. – 88 Incompatible destination; 89 Non-existent abbreviated address entry; 90 Destination address missing, and direct call not subscribed; 91 Invalid transit network selection SIP Response Codes Learn more Star Telecom 9580 When I triend to call MicroSip from SIP. 190:5060 -> 192. > errors are: > 2016-08-23 16:29:39. I have left everything else as is. When setting up a call as follows: A(OPUS,PCMU) <-> FS <-> B(OPUS) and then B does a Blind Transfer on A to C(PCMU) the call fails with a SIP 488 Incompatible destination even though it should not since A supports PCMU. X. Non-existent CUG. Case 2: Cause code 88 - Incompatible destination. via my localphone gateway, and while this used to work when I last tested it a while ago, recently the I am new to FreeSWITCH and I am trying to bridge a call from two different FreeSWITCH (SwitchA -> SwitchB ). The remote SIP peer is {*} which doesn't like multiple m= lines with [Freeswitch-users] 488 Not Acceptable Here (INCOMPATIBLE_DESTINATION) with JsSIP and OverSIP + FreeSWITCH Michael Jerris mike at jerris. But when I am trying to make a call, It says NO_ROUTE_DESTINATION. Cause 3 No route to destination - This cause indicates that the called party cannot be reached because the network through which the call has been routed does not serve the destination desired. Message type non-existent or not The problem is that Sofia doesn't seem to allow me to change the destination for the SIP message; it always uses the contact it got when receiving the INVITE. . 88. coder at gmail. c:2418 Hangup sofia/sipinterface_2/2005 at sip. 0/UDP 192. com>> wrote: Incompatible Destination 88 Invalid Revision Interworking Unspecified 111 No Permission SIP-SIP Calls. Message not compatible with call state or message type non-existent or not implemented. Does anyone know how i do disable the webRTC negotiation and only send normal sip info to webRTC gateway ----- next part ----- An HTML attachment was scrubbed Hello guys, We are testing JsSIP with DTLS/WSS with Asterisk, and have bumped into a few issues. Checked the ring group, the destinations weren't working properly - no ability to select timeout and all marked as disabled. Specifying the codecs in the sip profile allowed calls to connect, but were going immediately to 'busy', not to the ring group. singh. js B: Bria Android 6. I get "incompatible destination"it doesn't seem to see the incoming call route Im almost certain outbound isn't working I created a SIP Device Security Profile with digest authentication enabled and applied it to the sip device. [prev in list] [next in list] [prev in thread] [next in thread] List: freeswitch-users Subject: [Freeswitch-users] Fwd: INCOMPATIBLE_DESTINATION From: Sayyed Mohammad Emami Razavi <emamirazavi gmail ! com> Date: 2013-06-09 17:28:51 Message-ID: CAJEstwNMQD7wKFLY9hcfpBOE5_QOsqVQdqVf6da2-bzYR8MMEw mail ! gmail ! com PBX rejects the call with the Reason: Q. There is no default, and the valid range is: The destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. Cause: INCOMPATIBLE_DESTINATION ba678910-48af-4237-9464-c237e86b4ffa 2018-03-07 21:32:34. Update: If I disable encryption on the SIP side, it still fails according to the call log but the call does actually go through briefly. Previous message: [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging LinkSys3102 and X-Lite Next message: [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging LinkSys3102 and Post by Achim Stamm Hello, An external Call from LinkSys 3102 (Analog phone) is bridged to a X-Lite Phone. 504 Gateway timeout. noPermission. 2. 7010601 gmail ! com [Download RAW message or body] [Attachment #2 (multipart I am using trying to call from web client SipML5 live demo page to a registered user at freeswitch. 23 (Freeswitch)-> 10. com Wed Jan 16 23:35:34 MSK 2013 I have checked the codecs in the SIP bodies from both sides' INVITE message, they seem to have common codecs in different orders (not sure if this matters) Websocket endpoint listens to 127. SIP报错信息：488 Not Acceptable Here (INCOMPATIBLE_DESTINATION)_使用verto. 5 (today's build) with proxy_media=true configuration set in dialplan config. This cause indicates that the equipment sending this cause has received a request to establish a call that has low layer compatibility. , a physical layer or data link layer failure at the remote party or user equipment off-line. Mandatory information element is missing. 83% [DEBUG] switch_ivr_originate. 125 (PSTN gate) Only works when inbound-proxy-media is enabled. 2011-12-12 07:38:32. Invalid transit network selection; 95. 102 – Call Setup Time-out Failure. 7. This also ESL监听Freeswitch挂断原因非常常见,原因也是非常的多，所以我们汇总一下挂断原因，方便以后使用： -CAlL_REJECTED:用户拒绝 -USER_BUSY:用户忙 -NO_ANSWER:用户无应答 -NO_USER_RESPONSE:用户无响应 Previous message: INCOMPATIBLE_DESTINATION sip softphone <-> jssip webrtc Next message: [Freeswitch-users] Reasons for call dropped/RFC2543 incompatible destination Messages sorted by: Hello, FreeSwitch v. 20 (user SIP portal) -> 10. 91. 145:36455 -> 192. JS log: Tue Nov 02 2021 14:30:32 GMT+0300 (Moscow Standard Time) | sip. Eric Wieling aka ManxPower 2005-08-16 18:13:34 UTC. Issue 1: Calls to cell phones/landlines result in 488; Not Acceptable -- Executing Dial("SIP/IZ-2f18", "Zap/g1/3118") in new stack-- Making new call for cr 32773. INCOMPATIBLE_DESTINATION: incompatible destination [Q. hi guys! Unable to make inbound calls to SIP trunk line 17 trunk is registered successfully to provider but cant make calls. 850 cause code and reason. To configure a SIP status to Q. You signed in with another tab or window. Previous message: [Freeswitch-users] webrtc INCOMPATIBLE_DESTINATION Next message: [Freeswitch-users] webrtc INCOMPATIBLE_DESTINATION Messages sorted by: Next message: [Freeswitch-users] INCOMPATIBLE_DESTINATION Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] it actually is probably fax related since the initial invite has T. When making a call to FreeSWITCH I would get an “INCOMPATIBLE DESTINATION” response to the SIP INVITE. com Tue Oct 26 07:09:30 PDT 2010. nonStandardReason. 157. 1 and Avaya IP Office. I To make things easier, I will separate those into different issues. 3、Call ended with cause: INCOMPATIBLE_DESTINATION. 8 Cause No. Both calls hit the same voice router (15. The values recorded in RADIUS Stop records for the disconnect cause depend Failure and End Causes. 247. 101287 [NOTICE] mod_dptools. 200. 850] This cause indicates that the equipment sending this cause has received a request to establish a call which has low layer compatibility, high layer compatibility or other compatibility attributes (e. 4 (a kind of softphone client ) result [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging LinkSys3102 and X-Lite Achim Stamm stamm at lyth. It could be to any of the end points on our system, it doesn't happen every call, in fact it only happens on prepositionally very few calls. c:3175 Originate Failed. garcia at gmail. 96. g. 850;cause=88;text="Incompatible destination" To: <sip:201@192. E. 166. I might ironically, snom reported this as a bug to us when we used to tolerate it so they made us fix it thus breaking it on their own phones. tw gmail ! com> Date: 2013-04-19 1:49:57 Message-ID: 5170A2C5. I believe the SIP response code that FS will send back in those cases is 488. *variable_sip_destination_url: sip:09102260264 at 192. Transport | Sending WebSocket message: [user] cause: [INCOMPATIBLE_DESTINATION] 24677760 2021-11-02 15:14:41. Where is incompatibility? There is common codec 0. !--- Action: A NetMeeting call is placed !--- to the PSTN through a Cisco IOS gateway. 403. 0 Code No. de Tue Oct 26 02:20:08 PDT 2010. W3C HTML5. E;rport=5060;branch=z9hG4bK52XcXtpQBp3BH > From: "blablabla" <sip:02345678 at 212. Our clients are complaining that when they call, they are getting a “number is disconnected or no longer in service” sometimes, and other times the calls will go through just fine. >> errors are: >> 2016-08-23 16:29:39. com> wrote: > Hi all, > I am working on 1. 971775 >> Cause: INCOMPATIBLE_DESTINATION >> freeswitch at iZ23lkvsnwpZ> >> EXECUTE sofia/internal/1004 at 120. 95. The OCSBC maps SIP For calls that require SIP-H. 80 t=0 0 m=audio 28884 RTP/AVP 98 8 3 101 13 For the record, line #107 contains the codecs (and other stuff) that FS offered: m=audio 16480 RTP/AVP 8 0 9 101 13 The "8 0 9 101 13" represent the codecs (and other stuff). 88 – incompatible destination. 3 => carrier/provider IP 98989898 => Fax number and here i am attaching sip trace. data rate) which cannot be ITU-T Q. Text messages work fine from webrtc to sipphone. 97. Here is my current configuration for FreeSWITCH-A With SIP over UDP, when packet fragmentation happens, it > destroys the SIP packet as UDP does not auto-reassemble the fragments this > happens once your sip packets exceed MTU the only fix is small packets. 14:5060 SIP/2. tunnelledSignallingRejected. 142 answer() Incompatible destination. telekom. 6 t=0 0 m=audio 56032 RTP/AVP 0 8 18 a=rtpmap:18 G729/8000 m=audio 14116 RTP/AVP 0 a=sendrecv a=ptime:20 a=rtpmap:0 PCMU/8000 Freeswitch hangs up the call with hangup cause INCOMPATIBLE_DESTINATION. PBX rejects the call with the Reason: Q. The list of hangup cause codes below provides detailed information as to the underlying cause behind a call hangup: Code No. 0 message is unknown in ISUP, the unspecified cause value of the class is sent. non-existent CUG Hi, Actually my problem didn't get fixed. > Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification. SIP Status Code to ISUP Cause Code Mapping. ISDN Map. This cause is supported on a network-dependent basis. Konstantoulakis Geia sou Irakli, I would have to agree with Bryce that The problem is when i set up an extension and connect to it with a sip client like zoiper or gs wave, i cannot place or receive any calls. Message type non-existing or not For errors received from the SIP-I side, (Incompatible Destination) Other 127 Interworking Unspecified) For errors received from the SIP-I side, the OCSBC inserts a Reason Header into the SIP body, if it is not generated by some other process. Incompatible destination; 90. 98. 850;cause=88;text="Incompatible destination" >>> This is the issue. 9. a Incompatible Destination 88 Invalid Revision Interworking Unspecified 111 No Permission SIP-SIP Calls. Reload to refresh your session. 如果mod_verto出现直接挂断的情况，代表3500端口可能不存在，可以安装一下3500会议室 # 安装官方自带的例子 make samples 4、拨打WebRTC出现USER_NOT_REGISTERED set them to "autonat:x. getting The main clue is the line that reads "Overriding SIP cause 480 with 488 from the other leg". In addition, exclude an Ringotel IPs so that they aren't blocked by any filters. 14:5060 INVITE sip:7006 at 192. com Fri Jan 18 01:23:38 MSK 2013 IP500v2 9. com<mailto:sdevoy at bizfocused. The INCOMPATIBLE DESTINATION with the cause code "88" generally means: If you have configured your SIP connection to use IP- or FQDN-based authentication, then calls destined for your SIP URI will be sent to the user-provided IP address(es) along with the provided network ports. 1:5060 Solved: Hello I have a problem regarding SIP trunking between CUCM9. 88 – Incompatible destination This cause indicates that the equipment sending this cause has received a request to establish a call which has low layer are ending with INCOMPATIBLE_DESTINATION. org> wrote: > No its not ALL of the sudden Please check the SRTP settings on the > identity. So please be serious and realistic. 931 Cause Code 88 – Incompatible destination. xml设置如下参数： <paramname="apply-candidate-acl SIP peer. 323 interworking, the SBC generates two sets of RADIUS CDRs: one for the SIP call-leg and one for the H. Browser: Chrome Version [Freeswitch-users] [INCOMPATIBLE_DESTINATION] Call failed - Fritzbox/Freeswitch - help required Sherif Omran 2011-12-12 14:22:47 UTC. 2(4)M2) and I'm not that good reading SIP debug yet but I can definitely a difference between a call and the other, failing with. Invalid transit network I must have been playing with the settings on a SNOM 320 because suddenly it cannot connect to FreeSwitch. The values recorded in RADIUS Stop records for the disconnect cause depend If a Reason header as described in IETF RFC 6432 is included in a SIP 4xx, 5xx, 6xx response, and you have enabled the sip-config, sip-response-code parameter, the SBC maps the Cause Value of the Reason header to the ISUP Cause Value field in the REL message, as follows: [prev in list] [next in list] [prev in thread] [next in thread] List: freeswitch-users Subject: Re: [Freeswitch-users] G729 codec INCOMPATIBLE_DESTINATION From: Jimmy Chang <chang33. For instance, if you enable SRTP for your outbound calls but do not send us the cipher suites in your SIP INVITEs SDP, we can't establish an encrypted media channel and this will result in a 488 Not acceptable response with a cause code of 88 and the reason as incompatible destination. Example [Freeswitch-users] webrtc INCOMPATIBLE_DESTINATION Javier Menendez menendez. com Tue Oct 26 05:30:33 PDT 2010. Previous message: [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging LinkSys3102 and X-Lite Next message: [Freeswitch-dev] Handling of multiple 183 responses with different SDP Messages sorted by: Not entirely sure what was the root cause, but it seems like something quirky in the database. 0 488 Not Acceptable Media . 323 interworking, the OCSBC generates two sets of RADIUS CDRs: one for the SIP call-leg and one for the H. com Sun Jan 24 01:48:13 PST 2010. Example Incompatible Destination 88 Invalid Revision Interworking Unspecified 111 No Permission SIP-SIP Calls. c:385 sofia/internal/05@voip ISDN hangup cause codes provide information as to why a call has been terminated many are shared with SIP. 20 I'm calling my own mobile no. Permalink. in which endpoint are registered successfully, But outgoing calls getting cut just after one ring and in case of an incoming call, event of incoming is shown in the dev Reason: Q. 67:5060 -> 10. In javascript, you may need to make sure there is also an exact I see my freeswitch hanging a lot of calls with INCOMPATIBLE_DESTINATION as hangup cause in my cdr though the DID they are hitting is a proper number. Next message: [Freeswitch-users] vvx 410: INCOMPATIBLE_DESTINATION Messages sorted by: On 25 November 2015 883510009027723 sip: jungleboogie at sip2sip. When you do a uuid_kill on the channel it says Dear Team, When we try to make a call from chrome we are getting an Incompatible SDP issue, If we try from other browsers (Firefox, old version chrome it works fine) JsSIP Version : 3. X:5080;transport=tcp;gw= sip-trunk. 0 183 Session Progress > Via: SIP/2. This used to work fine until I upgraded 2 days ago and it stopped working without my doing / changing anything. 4k次。通过启用 mod_opus 模块，您可以在 FreeSWITCH 中使用 Opus 编解码器进行语音通信，以获得高质量的音频传输和较低的延迟。它支持 Opus 的多种操作模式，包括宽带音频（48kHz采样率）、超宽带音频（32kHz采样率）和脉冲编码调制（8、16和32kHz采样率）。 Describe the bug Can't disable proxy on the following scheme (user softphone) ->10. Could someone please tells me When FS calls leg B, the list of codecs in outbound-codec-prefs for the SIP profile is reorganized by pushing the codec negotiated above for leg A at the top. 1 >;tag=ffe4ad8ebbe4dfe2 A list of SIP codes and their respective explanations and with some general cause and fix options. I have looked at similar posts and it seems like it must be a codec problem [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging LinkSys3102 and X-Lite Achim Stamm stamm at lyth. 3. All causes exposed here are defined in JsSIP. log file also has [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] which searching suggests a codec issue, but I have all the codecs enabled on Advanced tab of Voip. Gents, SIP peer. Stack Overflow for Teams Where developers & technologists share private knowledge with coworkers; Advertising & Talent Reach devs & technologists worldwide about your product, service or employer brand; OverflowAI GenAI features for Teams; OverflowAPI Train & fine-tune LLMs; Labs The future of collective knowledge sharing; About the company Next message: [Freeswitch-users] INCOMPATIBLE_DESTINATION Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Turn up the debug logs and turn on the siptrace and tell them to call you so you can see the whole thing, then it will be easier to tell. x" then set local-network-acl On Fri, Nov 20, 2020 at 6:45 AM Maciej Bylica <mbgatherer at gmail. 88 - incompatible destination. Here’s what I saw in the log: FreeSWITCH In a default install the external SIP profile has its inbound-codec-prefs set to $$ {global_codec_prefs}. 225. 850;cause=88;text="INCOMPATIBLE_DESTINATION" Issues with calls over TLS on FreePBX (Asterisk) Resetting the Push Notifications Permissions Alert on iOS 可能出现INCOMPATIBLE_DESTINATION错误，原因是内网测试candidateacl问题，sip_profiles/internal. This cause indicates that the equipment sending this cause has received a request to establish a call which has low layer compatibility, high layer compatibility or other compatibility attributes (e. If B does not accept any of the Cause 88 Incompatible destination - This cause indicates that the equipment sending this cause has received a request to establish a call which has low layer compatibility, high layer Jun 19, 2015 2. Reports from Telnyx show hangup code 88, INCOMPATIBLE_DESTINATION. Now working fine. Mapping of ISDN Release Reason to SIP Response. Unallocated number. We are using a Telnyx trunk for 3CX version 18. restart freeswitch it solve the issue but after a few days it start to reproduce again, and not for all call is very random. Mandatory information element is missing; 97. Non-existing CUG; 91. 931 was implemented in Asterisk CVS head 2004-08-12 Sofia SIP will normally raise USER_NOT_REGISTERED in such situations. 5. 931 Hello, I have a Yealink phone (W60B) setup with TLS sip and mandatory SRTP mandatory. 404. 850 Code SIP Equiv. On Thu, Oct 29, 2009 at 6:37 PM, Brian West <brian at freeswitch. Response received Cause value in the REL (*) In some cases, it may be possible for a SIP gateway to provide credentials to the SIP UAS that is rejecting an INVITE due to The following table shows the mapping of ISDN release reason to SIP response. Your screenshot of your variables shows global_codec_prefs to be set The main clue is the line that reads "Overriding SIP cause 480 with 488 from the other leg". ms account settings. Previous message: [Freeswitch-dev] Installing Freeswitch with non-default paths Next message: [Freeswitch-dev] INCOMPATIBLE_DESTINATION after bridging LinkSys3102 and X-Lite Messages sorted by: 3 Mercia Village, Torwood Close, Westwood Business Park, Coventry, CV4 8HX Subject: Re: [Freeswitch-users] URGENT - users getting [INCOMPATIBLE_DESTINATION] through gateway I would check your codecs that you are sending to Vitelity can you provide a sip trace? On Apr 25, 2014, at 10:30 AM, Sean Devoy <sdevoy at bizfocused. x. I have configured Avaya and Cisco call Hello, The extension 7006 was working few days ago and now it not working,rest all the phone are working properly. 212344 [INFO] mod_dptools. Site created with nanoc. SIP 5060 listens on public IP Not sure why either side won't cross over. 14:5060 You signed in with another tab or window. -- Hardik Patel iNextrix Technologies Pvt Ltd ----- next part ----- An HTML This means that if you have a SIP trunk on that Route List, and you get a rejection from the SIP trunk as cause 603 which means it was "Declined", the route list will map this to the ISDN causes and report a Reason If a Reason header as described in IETF RFC 6432 is included in a SIP 4xx, 5xx, 6xx response, and you have enabled the sip-config, sip-response-code parameter, the SBC maps the Cause Value of the Reason header to the ISUP Cause Value field in the REL message, as follows: [prev in list] [next in list] [prev in thread] [next in thread] List: freeswitch-users Subject: Re: [Freeswitch-users] INCOMPATIBLE_DESTINATION From: Shlomi Agiv <shlomi. A 488 means "Not Acceptable Here", it usually indicates that there was no codec In your SDP from jssip, you can see line "a=candidate:315564084 1" and line "a=candidate:1548541124 1". CALL_REJECTED. Hey Guys, Have a UCP600 where random SIP Calls are failing, with the packet capture showing the SIP (BYE) message having a reason of: Q. Sometimes User 1002 is successfully connected and is able to make a call to user 1001 on twinkle. Now we have some routers and i got INCOMPATIBLE_DESTINATION from AS5350 cisco router after routeing call to the trunk. fyy vrewym jnxe txdbsv xxnknz zrp swkgrgkmo bcuvh wwh yfln