SIP: Can't dial phone number with pattern (XXX) XXX-XXXX
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Describe your environment
Please specify the following:
- OS: Guix System
- Jami version: 20230323.0
- What build you are using: Guix
Steps to reproduce
Note: Better the scenario is, better we will be able to reproduce and debug.
- Can you reproduce the bug: [at will | occasionally | not at all]
- Steps:
- Setup a SIP account 2. Try dialing a number in the (XXX) XXX-XXXX format
- Actual result: The call is immediately marked as failed
- Expected result: The call should proceed.
Workaround
The parenthesizes can be removed from the number, as well as the spaces or hyphens (XXXXXXXXXX) to make it work.
Additional information
Client logs:
[1691769528.394|9154|manager.cpp :1015] try outgoing call to '(514)768-1131' with account 'a2cf6bfa417527a8'
[1691769528.394|9154|sipaccount.cpp :164 ] [Account a2cf6bfa417527a8] Calling SIP peer (514)768-1131
[1691769528.394|9154|sipcall.cpp :144 ] [call:7391669446219807] Create a new [OUTGOING] SIP call with 1 media
[1691769528.394|9154|audio_rtp_session.cpp :58 ] Created Audio RTP session: 0x4bbb7b0 - call Id 7391669446219807
[1691769528.394|9154|ringbuffer.cpp :55 ] Create new RingBuffer 7391669446219807
[1691769528.394|9154|sipcall.cpp :2004] [call:7391669446219807] Added media @0: type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [camera://046d_HD_Pro_Webcam_C920_79D3DE2F] secure [YES]
[1691769528.394|9154|sipcall.cpp :2010] [call:7391669446219807] Created 1 Media streams
[1691769528.395|9154|sipcall.cpp :456 ] [call:7391669446219807] Setting transport to [0x7f0ab462a730]
[1691769528.395|9154|sipaccount.cpp :204 ] UserAgent: New registered account call to (514)768-1131
[1691769528.395|9154|ice_transport.cpp :333 ] [ice:0x6329d40] Creating IceTransport session for "7391669446219807"
[1691769528.395|9154|sipcall.cpp :3312] [call:7391669446219807] Successfully created media ICE transport [ice:0x3ef6440]
[1691769528.395|9154|sipcall.cpp :3474] [call:7391669446219807] Setting ICE session [0x3ef6440]
[1691769528.395|9154|sipcall.cpp :3334] [call:7391669446219807] Init media ICE transport
[1691769528.395|9154|ice_transport.cpp :408 ] [ice:0x6329d40] Initializing the session - comp count 2 - as a master
[1691769528.395|9154|ice_transport.cpp :449 ] [ice:0x6329d40] Add host candidates
[1691769528.395|9154|ice_transport.cpp :908 ] [ice:0x6329d40] added host stun config for UDP transport
[1691769528.395|9154|ice_transport.cpp :908 ] [ice:0x6329d40] added host stun config for UDP transport
[1691769528.396|9154|ice_transport.cpp :1018] [ice:0x6329d40] Missing local address, generic srflx candidates wont be generated!
[1691769528.396|9154|ice_transport.cpp :476 ] [ice:0x6329d40] No server reflexive candidates added
[1691769528.399|9154|ice_transport.cpp :709 ] [ice:0x6329d40] UDP initialization success
[1691769528.399|9154|ice_transport.cpp :772 ] [ice:0x6329d40] as master
[1691769528.399|9154|ice_transport.cpp :883 ] [ice:0x6329d40] (local) ufrag=73e64f01, pwd=78cf13ea6e4eb5a960b2e1df
[1691769528.400|9154|sdp.cpp :556 ] Creating SDP offer with 1 media
[1691769528.400|9154|sdp.cpp :264 ] Add media description [type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES]]
[1691769528.400|9154|sdp.cpp :504 ] [SDP OFFER] Local session (initial):
v=0
o=hurd 3900758328 0 IN IP4 192.168.10.15
s=Call ID 7391669446219807
c=IN IP4 192.168.10.15
t=0 0
m=audio 30902 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:30903 IN IP4 192.168.10.15
a=sendrecv
[1691769528.400|9154|manager.cpp :603 ] ----- Switch current call id to '7391669446219807' -----
[1691769528.400|9164|sipcall.cpp :1872] [call:7391669446219807] Add local attributes for ICE instance [0x3ef6440]
[1691769528.401|9164|sipcall.cpp :1912] [call:7391669446219807] add ICE local candidates for media [type [AUDIO] enabled [YES] muted [NO] label [audio_0]] @ 0
Removed sink: 0x03488480 from subscribers for id: 162056490676987
Creating new FrameObject for id: "7391669446219807"
Added sink: 0x03488480 to subscribers for id: 7391669446219807
[1691769528.401|9164|sipaccount.cpp :356 ] contact header: <sips:179835_desktop@192.168.10.15:47939;transport=tls> / <sips:179835_desktop@montreal6.voip.ms;transport=TLS> -> <sips:(514)768-1131@montreal6.voip.ms;transport=TLS>
[1691769528.401|9164|sipaccountbase.cpp :82 ] Creating SIP dialog:
From: <sips:179835_desktop@montreal6.voip.ms;transport=TLS>
Contact: <sips:179835_desktop@192.168.10.15:47939;transport=tls>
To: <sips:(514)768-1131@montreal6.voip.ms;transport=TLS>
[1691769528.401|9164|sipaccountbase.cpp :93 ] No target provided, using 'to' as target
[1691769528.401|9164|sipcall.cpp :744 ] [call:7391669446219807] Set new invite session [0x7f0aa8765858]
[1691769528.401|9164|sip_utils.cpp :220 ] Add header to SIP message: "User-Agent: Jami Daemon 13.7.0 (linux)"
[1691769528.402|9164|sipvoiplink.cpp :892 ] [call:7391669446219807] INVITE@0x7f0aa8765858 state changed to 1 (CALLING): cause=0, tsx@0x7f0b102ad818 status 0 (Default status message)
[1691769528.402|9164|call.cpp :241 ] [call:7391669446219807] state change 0/1, cnx 0/2, code 0
[1691769528.402|9164|call.cpp :275 ] [call:7391669446219807] emit client call state change CONNECTING, code 0
"slotCallStateChanged (call: 7391669446219807), from Recherche en cours to En cours de connexion"
[1691769528.489|9167|sipvoiplink.cpp :892 ] [call:7391669446219807] INVITE@0x7f0aa8765858 state changed to 6 (DISCONNCTD): cause=404, tsx@0x7f0ab494e3f8 status 404 (Not Found)
[1691769528.489|9167|call.cpp :241 ] [call:7391669446219807] state change 1/5, cnx 2/0, code 404
[1691769528.490|9167|call.cpp :275 ] [call:7391669446219807] emit client call state change FAILURE, code 404
[1691769528.490|9164|manager.cpp :1993] [call:7391669446219807] Parent call failed
[1691769528.490|9164|manager.cpp :1653] [call:7391669446219807] Remove local audio
[1691769528.490|9164|ringbufferpool.cpp :262 ] Unbind call 7391669446219807 from all bound calls
[1691769528.490|9164|sipcall.cpp :1589] [call:7391669446219807] removeCall()
[1691769528.490|9164|call_factory.cpp :72 ] Removing call 7391669446219807
[1691769528.490|9164|call_factory.cpp :75 ] Remaining 0 call
[1691769528.490|9164|call.cpp :241 ] [call:7391669446219807] state change 5/6, cnx 0/0, code 0
[1691769528.490|9164|call.cpp :275 ] [call:7391669446219807] emit client call state change OVER, code 0
[1691769528.490|9164|sipcall.cpp :730 ] [call:7391669446219807] Delete current invite session
[1691769528.491|9164|sipcall.cpp :456 ] [call:7391669446219807] Setting transport to [(nil)]
[1691769528.491|9164|audio_rtp_session.cpp :229 ] [0x4bbb7b0] Stopping receiver
[1691769528.491|9164|audio_rtp_session.cpp :68 ] Destroyed Audio RTP session: 0x4bbb7b0 - call Id 7391669446219807
[1691769528.491|9164|ringbuffer.cpp :60 ] Destroy RingBuffer 7391669446219807
[1691769528.491|9187|ice_transport.cpp :338 ] [ice:0x6329d40] destroying 0x5c6f298
"slotCallStateChanged (call: 7391669446219807), from En cours de connexion to Terminé"
"slotCallStateChanged (call: 7391669446219807), from Terminé to Terminé"
Invalid JSON: ""
[1691769528.900|9187|ice_transport.cpp :352 ] [ice:0x6329d40] Destroying ice_strans 0x5c6f298
[1691769528.902|9187|ice_transport.cpp :671 ] [ice:0x6329d40] Timer heap flushed after 0ms
[1691769528.902|9187|ice_transport.cpp :384 ] [ice:0x6329d40] done destroying
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