02-12-2018 11:33 AM - edited 03-21-2019 10:43 AM
Hi everyone,
I'm using a Cisco SPA 112 for a few months now without any problems. I recently changed my ISP and now I can't call one (quite usual) specific telephone number. Everything else works perfectly fine. I can call all other numbers I tried and receive all calls without problems. Even calls from the problematic number!
Additional information:
- Provider is Deutsche Telekom (Germany)
- latest firmware is installed
- number is reachable from other locations / devices
- I can reach the number from the mobile VoIP app of the provider
I hope someone can help me with this. :)
Here is the log file, when I try to call this number:
CSeq: 102 INVITE
Contact: <sip:sgc_c@217.0.27.52;transport=udp>
Supported: timer
Content-Length: 0
2018-02-09 18:37:35 Local0.Info 192.168.30.100
2018-02-09 18:37:35 Local0.Info 192.168.30.100
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 SIP_tsInviteClientEventProc(event:28)
2018-02-09 18:37:35 Local2.Debug 192.168.30.100 Start TmrD and send ACK
2018-02-09 18:37:35 Local0.Info 192.168.30.100 [0]->217.0.27.52:5060(773)
2018-02-09 18:37:35 Local0.Info 192.168.30.100 [0]->217.0.27.52:5060(773)
2018-02-09 18:37:35 Local7.Debug 192.168.30.100 ACK sip*********@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 192.168.30.100:5060;branch=z9hG4bK-bca60eeb
From: "Wohnzimmer" <sip:0049***********@tel.t-online.de>;tag=9c30c643d2ab193do0
To: <sip********@tel.t-online.de>;tag=h7g4Esbg_p65545t1518197846m429550c153299094s1_4216705964-598728892
Call-ID: b105973b-9d97bca5@192.168.30.100
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="0049***********",realm="tel.t-online.de",nonce="74FD12C462DC7D5A00000000CACB673D",uri="sip************@tel.t-online.de",algorithm=MD5,response="7580ed4e31264a6565f73a62a50c0baa",qop=auth,nc=00000001,cnonce="6475d35f"
Contact: "Wohnzimmer" <sip:0049*************@192.168.30.100:5060;ref=0049**************>
User-Agent: Cisco/SPA112-1.4.1SR1(002)
Content-Length: 0
2018-02-09 18:37:35 Local0.Info 192.168.30.100
2018-02-09 18:37:35 Local0.Info 192.168.30.100
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 SIP_sessTsEventProc(event:28)
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 xxxx SIP session.c b105973b-9d97bca5@192.168.30.100 processInviteResponse statusClass=13
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 SIP_releaseAudioResources() entered ################!!!!!!!!!!!!!!!!!
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 Requesting call statistics...
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 RTP TX stats updated for channel 0
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 RTP RX stats updated for channel 0
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 Call statistics updated.
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 AUD_releaseCallObj() call(0x1b21b8)
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 [AUD]AUD_stopRtpTx(0x1b21b8)
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 cordless_stop_rtp_tx(), Channel 0.
2018-02-09 18:37:35 Local0.Info 192.168.30.100 *** RTP channel not in Tx. Nothing to stop!
2018-02-09 18:37:35 Local0.Info 192.168.30.100 *** RTP channel not in Tx. Nothing to stop!
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 [AUD]RTP Tx Down
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 [AUD]AUD_stopRtpRx(0x1b21b8)
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 cordless_stop_rtp_rx(), Channel 0.
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 RTP channel 0 going from Rx to Idle.
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 RTP configuration:
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 audio_mode RTP_MODE_INACTIVE, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 Codec: duration 30, rx_pt_event 101, tx_pt_event 101, tx_pt 0
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 rx[0] 0 PCMU/8000, rx[1] 2 G.726/8000, rx[2] 8 PCMA/8000
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 rx[3] 18 G.729/8000, rx[4] 100 NSE/8000, rx[5] 112 encaprtp/8000
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 Jib: max 180ms, min 60ms, adapt 1
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 RTP channel 0 is now Idle.
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 [AUD]RTP Down
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 [AUD]AUD_releaseRtp(0x1b21b8)
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 cordless_stop_rtp(), releasing RTP channel:0
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 cordless_stop_rtp(), RTP session 0 stopped succussfully
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 uchRelChanAndEP(0, 3)
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 uchDisconnectEpFromNode(), disconnecting EP VoIP 0 from node 0
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 [AUD]RTP channel released
2018-02-09 18:37:35 Local2.Debug 192.168.30.100 [0:0]AUD Rel Call
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 SIP_releaseAudioResources(), CC_lineIsIdle(0)=0, gAudLine[0].bIvr=0, AUD_relUchNode????????????
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 SIP_releaseAudioResources() exit ################!!!!!!!!!!!!!!!!!
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 CC_eventProc(), event: CC_EV_SIG_CALL_FAILED(0x2A), lid: 0, par: 4, par2: 0x1a
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 AUD_ccEventProc: event 42 vid 0 par 0x4 par2 0x1a
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 callEventProcTable[3] is cepCallingProc
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 cepCallingProc(lid=0, call=0x17f9f4, event=42(CC_EV_SIG_CALL_FAILED), par=4, par2=0x1a)
2018-02-09 18:37:35 Local2.Debug 192.168.30.100 CC:Failed w/ Calling
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 NEW_CALL_STATE(), call 0: old state = CC_CST_CALLING, new state CC_CST_INVALID
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 SLIC_stopRing
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 SLIC_startTone 8
2018-02-09 18:37:35 Local3.Debug 192.168.30.100 ##### RTP_SEQ_NUM_EVT 14759
Solved! Go to Solution.
02-12-2018 03:18 PM
According the log, it seems the codec negotiation has failed. Unfortunately, log contain no SDP part of SIP packets, so no data to analyze. Capture SIP packets, please.
As a just blind shot - it seems SPA112 is not configured for Germany. It seems PCMU codec is allowed and preferred - it's good configuration for USA, suboptimal for Europe. Consider PCMA as most preferred (or only allowed if possible)( codec. Also, it seems you have RTP Packet Sice set to 30ms. It's good value nowhere - for both PCMA and PCMU codec. Use 20ms instead. Note that proposed changes may or may not solve the issue you are facing. I have no enough information to judge.
02-12-2018 03:18 PM
According the log, it seems the codec negotiation has failed. Unfortunately, log contain no SDP part of SIP packets, so no data to analyze. Capture SIP packets, please.
As a just blind shot - it seems SPA112 is not configured for Germany. It seems PCMU codec is allowed and preferred - it's good configuration for USA, suboptimal for Europe. Consider PCMA as most preferred (or only allowed if possible)( codec. Also, it seems you have RTP Packet Sice set to 30ms. It's good value nowhere - for both PCMA and PCMU codec. Use 20ms instead. Note that proposed changes may or may not solve the issue you are facing. I have no enough information to judge.
02-13-2018 05:39 AM
At first, thank you very much for taking the time to analyse the logfile.
I will change the settings today in the evening and tell you if it helped.
Which method do you reccommend to caputre the SIP packets of the SPA 112?
02-13-2018 06:07 AM
Binary form file produced by tcpdump or wireshark is most preferred (change file extension to .pcap.txt - you will not be allowed to attach file otherwise). SIP is text protocol, thus just plain text form is enough as well.
02-13-2018 09:14 AM
Hi,
you were right! After changing those values everything works great now.
Thank you so much!
02-13-2018 09:18 AM
I'm almost sure the 30ms of RTP Packet Size has caused it. Glad to hear you solved it.
Consider to claim comment with advice (not this one) correct answer - it will help others to found solutions.
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