06-15-2015 04:42 AM - edited 03-17-2019 03:21 AM
Hi,
I am configuring a SIP trunk on CME. In general this is quite simple, have done this before several times.
The SIP trunk is registered correctly, there is a dial-peer pointing to the SIP provider.
When I do a 'debug ccsip messages' I never see the SIP INVITE messages. After a lot of debugging the debug 'debug voip ccapi inout' tells me this:
Jun 15 14:32:12.199: //15346/FA871C57BAA2/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=1
Jun 15 14:32:12.199: //15348/FA871C57BAA2/CCAPI/cc_api_call_disconnected:
Cause Value=3, Interface=0x66238E88, Call Id=15348
The outgoing dial-peer is correct, but then we have an api_disconnect with cause code 3
Any idea?
Thanks
JH
06-15-2015 05:43 AM
Hi,
So far fro debugs, it seems that dial-plan was good, then next step is to establish connectivity to ISP to send the invite message. It seems that connection attempt fails , with remote server, so we are not able to see INVITE message itself in debugs.
- Are we using TCP or UDP as an session transport with ITSP.
Having more detailed debugs , along with configuration output will also help.
Debug ccsip message
debug ccsip info
debug ccsip error
or if there are not much calls on the CME, then you can also choose to run , " debug ccsip all " , to capture all output, please do not that logging rate should be disableed, Buffer set to large size.
06-15-2015 06:18 AM
06-15-2015 06:26 AM
Hello Jan,
Had a quick look, at options being printer, which we usually use as keep alive message for remote end:
Jun 15 16:11:36.613: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x6756A0D4, addr=217.10.79.9, port=5060, connId=2 for UDP Jun 15 16:11:36.613: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: OPTIONS sip:sipgate.de:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.200:5060;branch=z9hG4bK3F142402 From: <sip:192.168.2.200>;tag=20A5B9D8-1382 To: <sip:sipgate.de>
as we can see here that Options message is sourced from 192.168.2.200, is this ip address trusted by sipgate. Usually most of the telcos gives an WAN ip address for sip communications.
Also I would like to check , outgoing dial-peer status
show dial-peer voice summary
And last thing i would check, if CME is able to resolve FQDN to correct ip address:
ping sipgate.de
thanks,
Amit
06-15-2015 06:42 AM
Hi
I don't really understand the options send by CME, I would like to disable that. This is an older 2801 router, and I have not seen this before on CME. Don't think it is needed.
The sipgate.de is resolved correctly, I tried that several times.
The CME is behind a DSL router with SIP ALG enabled. The SIP registration works perfectly, when the SIP trunk re-registers, it gets an ACK back, so something seems OK:
Sent:
REGISTER sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.200:5060;branch=z9hG4bK3DE71576
From: <sip:4315124@sipgate.de>;tag=200A1FC0-1090
To: <sip:4315124@sipgate.de>
Date: Mon, 15 Jun 2015 10:21:40 GMT
Call-ID: C136E1B7-D9B11E5-8002F6DC-9B0F24A
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1434363700
CSeq: 2236 REGISTER
Contact: <sip:4315124@192.168.2.200:5060>
Expires: 3600
Authorization: Digest username="4315124",realm="sipgate.de",uri="sip:sipgate.de:5060",response="0a58137e4cfe1993a8b7b75924ad71eb",nonce="557ea98a6125922ff004d72273c1a154ace9f032",algorithm=md5
Content-Length: 0
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.200:5060;rport=64284;received=79.167.138.182;branch=z9hG4bK3DE71576
From: <sip:4315124@sipgate.de>;tag=200A1FC0-1090
To: <sip:4315124@sipgate.de>;tag=c3e497ecaece77a8e244e564b4212178.dd28
Call-ID: C136E1B7-D9B11E5-8002F6DC-9B0F24A
CSeq: 2236 REGISTER
Contact: <sip:4315124@192.168.2.200:5060>;expires=600;received="sip:79.167.138.182:64284"
Content-Length: 0
Only when we actually want to make a call we get this error code 3
SIP/Info/ccsip_set_cc_cause_for_spi_err: Categorized cause:3, category:128
Very weird
Jan
06-15-2015 06:57 AM
What we are seeing in dial-peer status command, as mentioned above, do we see dial-peer as up.
Options messages should have been enabled either in SIP-UA configuration or any outgoing dial-peer.
\Amit
06-15-2015 07:03 AM
Also SIP stack is throwing many message incomplete errors, where it says something received , but is unable to read that:
Received: ... Jun 15 16:11:18.143: //-1/xxxxxxxxxxxx/SIP/Error/HandleUdpIPv4SocketReads: SIP Message incomplete, trashed cme# Jun 15 16:11:27.610: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x66BB4710) with key=[15030] to table Jun 15 16:11:27.610: //15542/000000000000/SIP/State/sipSPIChangeState: 0x66BB4710 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE) Jun 15 16:11:27.610: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 37 Jun 15 16:11:27.610: //15542/000000000000/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 3CB6 to table Jun 15 16:11:27.610: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_DNS_RESOLVE cme# Jun 15 16:11:27.610: //15542/000000000000/SIP/State/sipSPIChangeState: 0x66BB4710 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_SENT_DNS) Jun 15 16:11:27.610: //15542/000000000000/SIP/State/sipSPIChangeState: 0x66BB4710 : State change from (STATE_IDLE, SUBSTATE_SENT_DNS) to (SIP_STATE_OPTIONS_WAIT, SUBSTATE_SENT_DNS) Jun 15 16:11:27.610: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: TYPE SRV query for _sip._udp.sipgate.de and type:1 cme# Jun 15 16:11:32.269: //-1/xxxxxxxxxxxx/SIP/Info/httpish_msg_process_network_msg: MSG LINE READ FAILURE DUE TO RS->EOF Jun 15 16:11:32.269: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_network_message: process_network_msg: not complete Jun 15 16:11:32.269: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: ... Jun 15 16:11:32.269: //-1/xxxxxxxxxxxx/SIP/Error/HandleUdpIPv4SocketReads: SIP Message incomplete, trashed cme#
Is there something else sitting in between CME & ITSP , like firewall etc. ??
Thanks,
Amit
06-15-2015 07:13 AM
Hi people,
Did you see this message:
Jun 15 16:11:32.269: //-1/xxxxxxxxxxxx/SIP/Info/httpish_msg_process_network_msg: MSG LINE READ FAILURE DUE TO RS->EOF
Verify the register timer as well:
registrar dns: sip.acme.com expires <60s between 3600s>
Best regards,
Daniel Sobrinho
06-17-2015 06:34 AM
For the people that were kind enough to help me:
I configured this on my own 2901 router and after some issues it worked.
The problem is that on older IOS routers (28xx) under sip-ua you can only put:
authentication username <username> password <your password>
In the newer 29XX routers I can put the commands:
credentials number <number> username <username> password <password> realm <realm>
authentication username <username> password <password>
And that made it work
And on top of this, this provider ONLY accepts g729
Shall try a little more on the 28xx router, but it seems this provider is not compatible with this way of configuring.
Thanks
Jan
06-17-2015 11:14 AM
Hi Jan,
Thank you for share +5
06-15-2015 07:03 AM
Hello Jan,
I agree with Amit. Check if CME is able to resolve FQDN to correct ip address.
3 – no route to destination = 404 Not Found
Share if possible the show runn config with us.
Best regards,
Daniel Sobrinho
06-15-2015 07:31 AM
Hello Daniel,
I did some searching and I found that this error is no route to destination. Although when I ping the FQDN, I get a reply. There is a small ADSL router in front, maybe this device does something ugly.
Can't contact customer now, but I shall contact again tomorrow.
Thank you for your help sofar
Bye
Jan
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