Pjsip syntax error exception when parsing request line header on line 1 col 1

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Pjsip syntax error exception when parsing request line header on line 1 col 1
Difficulties when connecting via Proxy #172
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Getting this error on a fully updated FreePBX system for one extension.

Anyone seen this before? Is this simply packet corruption?

Just confirmed that calls to this extension work, but calls from it do not.

We just might need to summon @JaredBusch to help you.

I’ve never had that error to my knowledge.

I just ran grep on the logs of a few PBX systems to check. Nothing found.

It’s my first time running into it. Wondering if it might not be a corrupted client.

Похоже, подключение к MangoLassi было разорвано, подождите, пока мы пытаемся восстановить соединение.

Источник

Difficulties when connecting via Proxy #172

Hello, I’m trying to install FreePBX inside a Portainer and I can make all connections to port 5060 normally, they all arrive at FreePBX but I’m having problems with the request headers (I’m testing via applications like Zoiper), see:

[2021-03-27 18:57:53] ERROR[1104] pjproject: sip_transport.c Error processing 597 bytes packet from UDP 192.168.100.1:39327 : PJSIP syntax error exception when parsing ‘Via’ header on line 2 col 37:
REGISTER sip:192.168.100.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:39327branch=z9hG4bK.99qn03bai;rport
From: sip:1000@192.168.100.100:5060;tag=MIRec9E0K
To: sip:1000@192.168.100.100:5060
CSeq: 35 REGISTER
Call-ID: f8x5CoJwbW
Max-Forwards: 70
Supported: replaces, outbound, gruu
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact: sip:1000@192.168.100.1:39327;transport=udp;+sip.instance=»urn:uuid:a601bec0-2f0d-00c4-8eeb-a908e24c9942″
Expires: 3600
User-Agent: LinphoneAndroid/4.3.1 (Mi A2) LinphoneSDK/4.4.2 (tags/4.4.2^0)

What happens externally is that a different machine, using Nginx, is trying to make a Proxy Pass for this one, apparently my Nginx is damaging the header (?), Is there any way to solve this?

Below my configuration on Nginx:

I don’t have much experience with this, I just tried to follow several tutorials on the internet and I was stuck here for a few days.

The text was updated successfully, but these errors were encountered:

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Не совсем понимаю, что означает и как полечить вот эту проблему. Догадываюсь, что где то в настройках пира, но не нахожу где.
То же самое подключение по SIP протоколу проходит успешно

[Apr  3 12:48:08] ERROR[6898]: pjproject:0 <?>:         sip_transport. Error processing 1038 bytes packet from UDP 1.1.1.1:5080 : PJSIP syntax error exception when parsing 'Authorization' header on line 13 col 200:
REGISTER sip:sip.mysip.com;transport=udp SIP/2.0
v:SIP/2.0/UDP 1.1.1.1:5080;rport;branch=z9hG4bKS0HQ7aQ905FHp
Max-Forwards:70
f:<sip:123456@sip.mysip.com>;tag=8aHX5Zg0pppZS
t:<sip:123456@sip.mysip.com>
i:637dcbbf-b33b-40d0-96e4-3a2086c060bb
CSeq:2568836 REGISTER
m:<sip:gw+3b305cad0286d1231d0633b417009d72@1.1.1.1:5080;transport=udp;gw=3b305cad0286d1231d0633b417009d72>
Expires:3600
User-Agent:onlinepbx s3
Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
k:path,replaces
Authorization:Digest username="123456",realm="sip.mysip.com",nonce="0a741c02",algorithm=MD5,uri="sip:sip.mysip.com;transport=udp",response="e6fc6915394e79079618a80a7751c591",Digest username="123456",realm="asterisk",nonce="1554242124/6bd062f852ef54a2083391ef532fbbf9",cnonce="3EFai9A0EjezAQAwSM7MLg",opaque="0689dba2653326ec",algorithm=MD5,uri="sip:sip.mysip.com;transport=udp",response="b4fc7e1bc8e7b3bad7e996f0190ea2a0",qop=auth,nc=00000006

pjsip show endpoint 123456

 ParameterName                      : ParameterValue
 ==============================================================
 100rel                             : yes
 accountcode                        : 99999999
 acl                                :
 aggregate_mwi                      : true
 allow                              : (ulaw|alaw|gsm|g729|opus)
 allow_overlap                      : true
 allow_subscribe                    : true
 allow_transfer                     : true
 aors                               : 123456
 asymmetric_rtp_codec               : false
 auth                               : 123456
 bind_rtp_to_media_address          : false
 call_group                         :
 callerid                           : 111111111
 callerid_privacy                   : allowed_not_screened
 callerid_tag                       :
 connected_line_method              : invite
 contact_acl                        :
 context                            : a2billing
 cos_audio                          : 0
 cos_video                          : 0
 device_state_busy_at               : 0
 direct_media                       : false
 direct_media_glare_mitigation      : none
 direct_media_method                : invite
 disable_direct_media_on_nat        : false
 dtls_ca_file                       :
 dtls_ca_path                       :
 dtls_cert_file                     :
 dtls_cipher                        :
 dtls_fingerprint                   : SHA-256
 dtls_private_key                   :
 dtls_rekey                         : 0
 dtls_setup                         : active
 dtls_verify                        : No
 dtmf_mode                          : auto_info
 fax_detect                         : false
 fax_detect_timeout                 : 0
 force_avp                          : false
 force_rport                        : true
 from_domain                        :
 from_user                          :
 g726_non_standard                  : false
 ice_support                        : false
 identify_by                        : username,ip
 inband_progress                    : false
 incoming_mwi_mailbox               :
 language                           : ru
 mailboxes                          :
 media_address                      :
 media_encryption                   : no
 media_encryption_optimistic        : false
 media_use_received_transport       : false
 message_context                    :
 moh_suggest                        : default
 mwi_from_user                      :
 mwi_subscribe_replaces_unsolicited : false
 named_call_group                   :
 named_pickup_group                 :
 notify_early_inuse_ringing         : false
 one_touch_recording                : false
 outbound_auth                      :
 outbound_proxy                     :
 pickup_group                       :
 record_off_feature                 : automixmon
 record_on_feature                  : automixmon
 refer_blind_progress               : true
 rewrite_contact                    : false
 rpid_immediate                     : false
 rtcp_mux                           : false
 rtp_engine                         : asterisk
 rtp_ipv6                           : false
 rtp_keepalive                      : 0
 rtp_symmetric                      : false
 rtp_timeout                        : 0
 rtp_timeout_hold                   : 0
 sdp_owner                          : -
 sdp_session                        : Asterisk
 send_diversion                     : true
 send_pai                           : false
 send_rpid                          : false
 set_var                            :
 srtp_tag_32                        : false
 sub_min_expiry                     : 0
 subscribe_context                  :
 t38_udptl                          : false
 t38_udptl_ec                       : none
 t38_udptl_ipv6                     : false
 t38_udptl_maxdatagram              : 0
 t38_udptl_nat                      : false
 timers                             : yes
 timers_min_se                      : 90
 timers_sess_expires                : 1800
 tone_zone                          :
 tos_audio                          : 0
 tos_video                          : 0
 transport                          : transport-udp
 trust_id_inbound                   : false
 trust_id_outbound                  : false
 use_avpf                           : false
 use_ptime                          : false
 user_eq_phone                      : false
 voicemail_extension                :

pjsip show aor 123456

 ParameterName        : ParameterValue
 =====================================
 authenticate_qualify : false
 contact              :
 default_expiration   : 3600
 mailboxes            :
 max_contacts         : 20
 maximum_expiration   : 7200
 minimum_expiration   : 60
 outbound_proxy       :
 qualify_frequency    : 0
 qualify_timeout      : 3.000000
 remove_existing      : false
 support_path         : false
 voicemail_extension  :

pjsip show auth 123456

 ParameterName  : ParameterValue
 ===============================
 auth_type      : userpass
 md5_cred       :
 nonce_lifetime : 32
 password       : my_strong_pass
 realm          :
 username       : 123456

спросил

2019-04-03 04:09:05 +0400

etskh
326 ● 40 ● 18

Источник

When using Kamailio from the master branch, I’m encountering an issue where IPv6 contact address aliases that are added via set_contact_alias() for WebSocket connections (for example) are unquoted and create problems for Asterisk:

pjsip:0 <?>:    sip_transport. Error processing 3633 bytes packet from UDP 10.0.0.1:5060 : PJSIP syntax error exception when parsing 'Request Line' header on line 11 col 129:

and TOPOH garbles the Request-URI when processing the ACK:

WARNING: sanity [sanity.c:236]: check_ruri_scheme(): failed to parse request 
uri [�a�{1me▒s�na@50�9���1.�8:�2v0;i��a��=11>7�n7x>10O�2v0~1]

I have tested this with CSipSimple nightly branch (PJSIP) Asterisk 13.2.0 (PJSIP), Google Chrome 41.0.2272.118, and Firefox 37 and all fail to handle the Contact alias parameter properly.

An example of an invite that shows this issue follows (this INVITE is without TOPOH on).

INVITE sip:testuser@example.com SIP/2.0
Record-Route: <sip:10.0.0.1;r2=on;lr=on;ftag=t5r2736hf9;nat=yes>
Record-Route: <sip:[2001:db8::98]:5061;transport=ws;r2=on;lr=on;ftag=t5r2736hf9;nat=yes>
Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKa3e8.2bb6508fbc0cf8cc55c1eb6c0eca0b38.0
Via: SIP/2.0/WSS 2e6orc23ptjv.invalid;rport=43691;received=2001:db8::99;branch=z9hG4bK450538
Max-Forwards: 69
To: <sip:testuser@example.com>
From: "WS Test User 1" <sip:wstest1@example.com>;tag=t5r2736hf9
Call-ID: uugmpjgjpoklrnf0um56
CSeq: 7244 INVITE
Contact: <sip:wstest1@example.com;gr=urn:uuid:2e54e8a2-66e4-433a-a024-b57f3665a44b;alias=[2001:db8::99]~43691~6>
Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE
Content-Type: application/sdp
Supported: gruu,outbound
User-Agent: SIP.js/0.6.4
Content-Length: 2768

v=0
o=- 867554869709517848 2 IN IP4 10.0.0.1
s=-
t=0 0
a=msid-semantic: WMS livWfUwMWQJMemgnFyBQDf1VXUo9Q0AXwHnH
m=audio 30158 RTP/SAVP 111 103 104 9 0 8 106 105 13 126
c=IN IP4 10.0.0.1
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:1444733772 cname:CuBdEZ9bIWQc1sE+
a=ssrc:1444733772 msid:livWfUwMWQJMemgnFyBQDf1VXUo9Q0AXwHnH ebd0066b-73b9-467c-b696-36e0fa7d72b7
a=ssrc:1444733772 mslabel:livWfUwMWQJMemgnFyBQDf1VXUo9Q0AXwHnH
a=ssrc:1444733772 label:ebd0066b-73b9-467c-b696-36e0fa7d72b7
a=sendrecv
a=rtcp:30159
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/fmyETo1BwAlupltb64sy5Za6e37BW0p5jMmvqHU
a=setup:actpass
a=fingerprint:sha-1 76:27:60:1E:64:94:B4:6E:8A:64:72:2D:41:2C:B8:F3:FF:4C:1D:56
a=ice-ufrag:U4XlaM4U
a=ice-pwd:HqnTPTU5DY3j1yB4XVEL9rs1tn
a=candidate:vVzo18e51E8EXCNy 1 UDP 2130706431 10.0.0.1 30158 typ host
a=candidate:vVzo18e51E8EXCNy 2 UDP 2130706430 10.0.0.1 30159 typ host
a=candidate:xCpWH2paVucFSQXn 1 UDP 2130706175 2001:db8::1 30158 typ host
a=candidate:xCpWH2paVucFSQXn 2 UDP 2130706174 2001:db8::1 30159 typ host
m=video 30196 RTP/SAVP 100 116 117 96
c=IN IP4 10.0.0.1
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
a=ssrc-group:FID 616715905 3177427003
a=ssrc:616715905 cname:CuBdEZ9bIWQc1sE+
a=ssrc:616715905 msid:livWfUwMWQJMemgnFyBQDf1VXUo9Q0AXwHnH bb5df7ec-28a9-4e94-89a3-2ebb45a9f5bb
a=ssrc:616715905 mslabel:livWfUwMWQJMemgnFyBQDf1VXUo9Q0AXwHnH
a=ssrc:616715905 label:bb5df7ec-28a9-4e94-89a3-2ebb45a9f5bb
a=ssrc:3177427003 cname:CuBdEZ9bIWQc1sE+
a=ssrc:3177427003 msid:livWfUwMWQJMemgnFyBQDf1VXUo9Q0AXwHnH bb5df7ec-28a9-4e94-89a3-2ebb45a9f5bb
a=ssrc:3177427003 mslabel:livWfUwMWQJMemgnFyBQDf1VXUo9Q0AXwHnH
a=ssrc:3177427003 label:bb5df7ec-28a9-4e94-89a3-2ebb45a9f5bb
a=sendrecv
a=rtcp:30197
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qQRmOwum0YFHgKKzt2dRX0uAQhSwlmxscI3P3JUI
a=setup:actpass
a=fingerprint:sha-1 76:27:60:1E:64:94:B4:6E:8A:64:72:2D:41:2C:B8:F3:FF:4C:1D:56
a=ice-ufrag:ISaptX08
a=ice-pwd:mxP343kHexU8SwtbCeRj4WyMNI
a=candidate:vVzo18e51E8EXCNy 1 UDP 2130706431 10.0.0.1 30196 typ host
a=candidate:vVzo18e51E8EXCNy 2 UDP 2130706430 10.0.0.1 30197 typ host
a=candidate:xCpWH2paVucFSQXn 1 UDP 2130706175 2001:db8::1 30196 typ host
a=candidate:xCpWH2paVucFSQXn 2 UDP 2130706174 2001:db8::1 30197 typ host

I believe that the Contact header alias parameter might need quotes in order to properly handle the bracketed IPv6 alias address, something like the following, but unfortunately I haven’t been able to find evidence to be certain that this is the fix, though I do notice that parameters in headers with special characters like brackets are often quoted.

Contact: <sip:wstest1@example.com;gr=urn:uuid:2e54e8a2-66e4-433a-a024-b57f3665a44b;alias="[2001:db8::99]~43691~6">

Источник

Pjsip syntax error exception when parsing request line header on line 1 col 1

welcome to the Polycom Community.

The SPIP550 should be on UC Software 4.0.11 and not 4.1.1 as this is for LYNC only.

I suggest you post your configuration and try and get some logs of the VVX Phone.

Settings > Logging > Global Log Level Limit > Debug
Settings > Logging > Log File Size (Kbytes) > 160
Settings > Logging > Module Log Level Limits > SIP > Debug
Settings > Logging > Module Log Level Limits > TLS > Debug
Settings > Logging > Module Log Level Limits > CURL > Debug

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Polycom Global Services

If official support is required please check how to phone or open a case here

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Please also ensure you always check the VoIP , Video Endpoint , Microsoft Voice , PSTN or other FAQ’s in the different sections

Источник

Pjsip syntax error exception when parsing request line header on line 1 col 1

Getting this error on a fully updated FreePBX system for one extension.

Anyone seen this before? Is this simply packet corruption?

Just confirmed that calls to this extension work, but calls from it do not.

We just might need to summon @JaredBusch to help you.

I’ve never had that error to my knowledge.

I just ran grep on the logs of a few PBX systems to check. Nothing found.

It’s my first time running into it. Wondering if it might not be a corrupted client.

Похоже, подключение к MangoLassi было разорвано, подождите, пока мы пытаемся восстановить соединение.

Источник

Pjsip syntax error exception when parsing request line header on line 1 col 1

welcome to the Polycom Community.

The SPIP550 should be on UC Software 4.0.11 and not 4.1.1 as this is for LYNC only.

I suggest you post your configuration and try and get some logs of the VVX Phone.

Settings > Logging > Global Log Level Limit > Debug
Settings > Logging > Log File Size (Kbytes) > 160
Settings > Logging > Module Log Level Limits > SIP > Debug
Settings > Logging > Module Log Level Limits > TLS > Debug
Settings > Logging > Module Log Level Limits > CURL > Debug

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Polycom Global Services

If official support is required please check how to phone or open a case here

Источник

Difficulties when connecting via Proxy #172

Comments

[2021-03-27 18:57:53] ERROR[1104] pjproject: sip_transport.c Error processing 597 bytes packet from UDP 192.168.100.1:39327 : PJSIP syntax error exception when parsing ‘Via’ header on line 2 col 37:
REGISTER sip:192.168.100.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:39327branch=z9hG4bK.99qn03bai;rport
From: sip:1000@192.168.100.100:5060;tag=MIRec9E0K
To: sip:1000@192.168.100.100:5060
CSeq: 35 REGISTER
Call-ID: f8x5CoJwbW
Max-Forwards: 70
Supported: replaces, outbound, gruu
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact: sip:1000@192.168.100.1:39327;transport=udp;+sip.instance=»urn:uuid:a601bec0-2f0d-00c4-8eeb-a908e24c9942″
Expires: 3600
User-Agent: LinphoneAndroid/4.3.1 (Mi A2) LinphoneSDK/4.4.2 (tags/4.4.2^0)

What happens externally is that a different machine, using Nginx, is trying to make a Proxy Pass for this one, apparently my Nginx is damaging the header (?), Is there any way to solve this?

Below my configuration on Nginx:

I don’t have much experience with this, I just tried to follow several tutorials on the internet and I was stuck here for a few days.

The text was updated successfully, but these errors were encountered:

Источник

Элтекс

производство оборудования для телекоммуникаций

SMG 2016 Анонимные вызовы Анти АОН anonymous@anonymous.invalid

Сообщение andrsharov » 17 сен 2020 18:19

Проблема: при входящем вызове на Eltex SMG 2016 от абонента с включенным антиопределителем номера, Eltex передает не все поля Contact в Header.

Вводные данные:
Оператор местной телефонной связи.
Eltex SMG-2016.
Версия ПО: ECSS-10 V.3.18.0.3969 2016/PBX/SORM/VAS/REC/IVR Build: Jul 20 2020 16:07:52
Присоединяющий оператор: Beeline
Схема трафика: Beeline SMG-2016 Asterisk(PJSIP) Абоненты .

Входящий вызов от Beeline в сторону SMG 2016 приходит со следующими полями

Message Header
Via: SIP/2.0/UDP 8.8.8.8:5060;branch=z9hG4bKfupkf53tnn74v979q9r344qrs;Role=3;Hpt=8e58_16;TRC=ffffffff-ffffffff
Record-Route:
Call-ID: isbcuoczkpzqzo4yhr1qosr114sqchzz4qfo@SoftX3000
[Generated Call-ID: isbcuoczkpzqzo4yhr1qosr114sqchzz4qfo@SoftX3000]
From: Anonymous ;tag=z140uyyp-CC-27
SIP Display info: Anonymous
SIP from address: sip:anonymous@anonymous.invalid;cpc-rus=1
SIP from address User Part: anonymous
SIP from address Host Part: anonymous.invalid
SIP From URI parameter: cpc-rus=1
SIP from tag: z140uyyp-CC-27
To:
CSeq: 1 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact:
Contact URI: sip:Anonymous@8.8.8.8:5060;transport=udp;Hpt=8e58_16;CxtId=4;TRC=ffffffff-ffffffff
Contact URI User Part: Anonymous
Contact URI Host Part: 8.8.8.8
Contact URI Host Port: 5060
Contact URI parameter: transport=udp
Contact URI parameter: Hpt=8e58_16
Contact URI parameter: CxtId=4
Contact URI parameter: TRC=ffffffff-ffffffff
Max-Forwards: 68
Supported: 100rel,timer
User-Agent: Huawei SoftX3000 V300R011
Session-Expires: 3600
Min-SE: 600
Privacy: user
Content-Length: 276
Content-Type: application/sdp
Message Body

Но далее выходит с не полным набором значений в поле Contact

Frame 1: 902 bytes on wire (7216 bits), 902 bytes captured (7216 bits)
Ethernet II, Src: EltexEnt_8b:4f:0a (a8:f9:4b:8b:4f:0a), Dst: c2:cc:da:c8:6a:24 (c2:cc:da:c8:6a:24)
Internet Protocol Version 4, Src: 77.88.8.8, Dst: 77.88.8.2
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:74957777777@77.88.8.2:5060;user=phone SIP/2.0
Message Header
Via: SIP/2.0/UDP 77.88.8.8:5060;rport;branch=z9hG4bK-o638868-1828
Transport: UDP
Sent-by Address: 77.88.8.8
Sent-by port: 5060
RPort: rport
Branch: z9hG4bK-o638868-1828
From: «Anonymous» ;tag=o1828p0D1719D0t7r638831
SIP Display info: «Anonymous»
SIP from address: sip:77.88.8.8;user=phone
SIP from address Host Part: 77.88.8.8
SIP From URI parameter: user=phone
SIP from tag: o1828p0D1719D0t7r638831
To:
SIP to address: sip:74957777777@77.88.8.2;user=phone
SIP to address User Part: 74957777777
SIP to address Host Part: 77.88.8.2
SIP To URI parameter: user=phone
Call-ID: 1600-332168-638677
[Generated Call-ID: 1600-332168-638677]
CSeq: 2 INVITE
Sequence Number: 2
Method: INVITE
User-Agent: smg_pa_sip 3.18.0.63
Max-Forwards: 29
Contact:
Contact URI: sip:@77.88.8.8:5060
Contact URI Host Part: 77.88.8.8
Contact URI Host Port: 5060
Accept: multipart/mixed, application/sdp
Allow: INVITE, ACK, BYE, CANCEL, PRACK, REGISTER, INFO, REFER, NOTIFY, OPTIONS, UPDATE
Supported: replaces
X-UniqueTag: 110000d15f632188365515439d437001
Content-Type: application/sdp
Content-Length: 236
Message Body
Session Description Protocol

Asterisk почему-то это поле Contact не нравится , и он в консоли он выдает ошибку:

[Sep 16 17:15:00] ERROR[601]: pjproject: : sip_transport.c Error processing 860 bytes packet from UDP 77.88.8.8:5060 : PJSIP syntax error exception when parsing ‘Contact’ header on line 9 col 15:

Т.е. с таким полем Contact все работает

А вот с таким полем Contact, работать не хочет:

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