Solved: Problem with registration Cisco 2801 as SIP gateway with SIP server

m.hyza · ‎04-16-2014

Hello, please help me with the configuration of CPE router as the Cisco 2801 SIP UAC. CPE router can not register to the SIP server (FXS phone. - Cisco 2801 - SIP server). In Wireshark I see that I get the Message Router SIP 401 Unauthorized.

Here is applicable configuration:

SIP#
...
voice service voip
no ip address trusted authenticate
allow-connections sip to sip
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface FastEthernet0/0.765
bind media source-interface FastEthernet0/0.765
early-offer forced
sip-profiles 1
...
sip-ua
credentials username +421906200200 password 7 ABC realm ABC.DEF.COM
authentication username +421906200200 password 7 ABC realm ABC.DEF.COM
no redirection
registrar dns:ABC.DEF.COM expires 3600
sip-server dns:ABC.DEF.COM
...

And here's the debug output from ccsip all:

SIP#show sip register stat
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
+421906200200 -1 102 no

SIP#debug ccsip al

*Apr 16 08:10:24.512: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x68CA80D8) with key=[736] to table
*Apr 16 08:10:24.512: //-1/000000000000/SIP/Info/ccsip_offer_ans_init:
*Apr 16 08:10:24.512: //-1/000000000000/SIP/Info/ccsip_iwf_init:
*Apr 16 08:10:24.512: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init:
*Apr 16 08:10:24.512: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization: Entry...
*Apr 16 08:10:24.512: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetSipProfilesTag: voice class SIP Profiles tag is set : 1
*Apr 16 08:10:24.512: //-1/000000000000/SIP/Info/ccsipRegisterSetTargetInfo: p2p mode with Registrar Server = dns:imspp.orange.sk
*Apr 16 08:10:24.512: //-1/000000000000/SIP/Info/ccsipRegisterSetTargetInfo: Parsing The Registrar Address
*Apr 16 08:10:24.512: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: target_host : imspp.orange.sk target_port : 5060

*Apr 16 08:10:24.512: //-1/000000000000/SIP/Info/sipSPIValidateAndCopyOutboundHost: CCSIP: copy target_host to outbound_host
*Apr 16 08:10:24.512: //-1/000000000000/SIP/Info/sipSPIOutboundProxyReuse: Do not reuse Outbound Proxy IP adress and Port
*Apr 16 08:10:24.512: //-1/000000000000/SIP/State/sipSPIChangeState: 0x68CA80D8 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
*Apr 16 08:10:24.512: //-1/000000000000/SIP/Info/ccsip_spi_registrar_add_expires_header: Inside ccsip_spi_registrar_add_expires_header for Expires
*Apr 16 08:10:24.512: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_OUTBOUND_REGISTER
*Apr 16 08:10:24.512: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIIncrementOverloadCount: Local 1 Global 1
*Apr 16 08:10:24.516: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 40
*Apr 16 08:10:24.516: //-1/000000000000/SIP/Info/act_idle_outgoing_register: In act_idle_outgoing_register

*Apr 16 08:10:24.516: //725/000000000000/SIP/Info/act_idle_outgoing_register: Se
SIP#nd REGISTER to imspp.orange.sk:5060

*Apr 16 08:10:24.516: //725/000000000000/SIP/Info/sipSPIUaddCcbToUACTable: ****Adding to UAC table.
*Apr 16 08:10:24.516: //725/000000000000/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x68CA80D8 key=45A45733-C3C611E3-800DB53B-FCD69C43
*Apr 16 08:10:24.516: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_DNS_RESOLVE
*Apr 16 08:10:24.516: //725/000000000000/SIP/State/sipSPIChangeState: 0x68CA80D8 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_SENT_DNS)
*Apr 16 08:10:24.516: //725/000000000000/SIP/State/sipSPIChangeState: 0x68CA80D8 : State change from (STATE_IDLE, SUBSTATE_SENT_DNS) to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_SENT_DNS)
*Apr 16 08:10:24.516: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: TYPE SRV query for _sip._udp.imspp.orange.sk and type:1
SIP#
*Apr 16 08:10:42.516: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_aaaa_query: DNS query for imspp.orange.sk and type:1
*Apr 16 08:10:42.516: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_query: TYPE A query successful for imspp.orange.sk
*Apr 16 08:10:42.516: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_query: ttl for A records = 0 seconds
*Apr 16 08:10:42.516: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_aaaa_query: IP Address of imspp.orange.sk is:

*Apr 16 08:10:42.516: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_aaaa_query: 213.151.230.248

*Apr 16 08:10:42.516: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 43
*Apr 16 08:10:42.516: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICacheHostToCCB: sipSPICacheHostToCCB dnsResponse.num_hosts = 1
*Apr 16 08:10:42.516: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICacheHostToCCB: IP Address No. 1, IP address 213.151.230.248
*Apr 16 08:10:42.516: //725/000000000000/SIP/Info/resolve_sig_ip_address_to_bind: signaling bind address : 192.168.49.6
*Apr 16 08:10:42.516: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_api_register_target_dns_resolved: ttl = 0
*Apr 16 08:10:42.516: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_spi_register_get_rcb: Getting New RCB [0x691D01B0]
*Apr 16 08:10:42.516: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_register_set_dns_resolved_address: CCSIP_REGISTER:: registrar 0 DNS resolved addr set to 213.151.230.248:5060
*Apr 16 08:10:42.516: //-1/xxxxxxxxxxxx/SIP/Info/ccsipRegisterStartRCBTimer: Starting timer for pattern for 3600 seconds
*Apr 16 08:10:42.516: //725/000000000000/SIP/State/sipSPIChangeState: 0x68CA80D8 : State change from (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_SENT_DNS) to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
*Apr 16 08:10:42.516: //725/000000000000/SIP/Info/sipSPIPresendProcessing: Presend Processing called for 7 event
*Apr 16 08:10:42.520: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIRetrieveOutgoingPassThruData: Retrieving Data from RCB
*Apr 16 08:10:42.520: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIRetrieveOutgoingPassThruData: Retrievi
SIP#ng Data from RCB
*Apr 16 08:10:42.520: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone METDST to SIP default timezone = GMT
*Apr 16 08:10:42.520: //725/000000000000/SIP/Info/sipSPISendRegister: Associated container=0x68EEFC2C to Register
*Apr 16 08:10:42.520: //725/000000000000/SIP/Transport/sipSPISendRegister: Sending REGISTER to the transport layer
*Apr 16 08:10:42.520: //725/000000000000/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
*Apr 16 08:10:42.520: //725/000000000000/SIP/Transport/sipSPITransportSendMessage: msg=0x680DA888, addr=213.151.230.248, port=5060, sentBy_port=0, local_addr=192.168.49.6, is_req=1, transport=1, switch=0, callBack=0x6181F574
*Apr 16 08:10:42.520: //725/000000000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Apr 16 08:10:42.520: //725/000000000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Apr 16 08:10:42.520: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:213.151.230.248, rport:5060 with laddr:192.168.49.6

*Apr 16 08:10:42.520: //725/000000000000/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x680DA888
*Apr 16 08:10:42.520: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x680DA888, addr=213.151.230.248, port=5060, local_addr=192.168.49.6, connId=3 for UDP
*Apr 16 08:10:42.520: //725/000000000000/SIP/State/sipSPIChangeState: 0x68CA80D8 : State change from (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE) to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
*Apr 16 08:10:42.524: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:
REGISTER sip:imspp.orange.sk:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.49.6:5060;branch=z9hG4bK2D496B
From: <sip:+421906200200@imspp.orange.sk>;tag=958E83C-1011
To: <sip:+421906200200@imspp.orange.sk>
Date: Wed, 16 Apr 2014 06:10:42 GMT
Call-ID: 45A45733-C3C611E3-800DB53B-FCD69C43
User-Agent: Cisco-SIPGateway/IOS
SIP#-12.x
Max-Forwards: 70
Timestamp: 1397628642
CSeq: 320 REGISTER
Contact: <sip:+421906200200@192.168.49.6:5060>
Expires: 3600
Supported: path
Content-Length: 0

Daniele Giordano · ‎05-03-2014

HI.

Questions and Answers:

1. To I solved 407 Proxy Authentication Required I should not be configured to "voice class sip-profiles 1" command "request INVITE sip-header Proxy-Authorization ???"

In my opinion the call flow is correct. Your cisco sends an INVITE without authentication and so the provider uses a 407 message to get a new INVITE with authentication parameters. This exchange is very fast and it doesn't add delay:
*May 3 11:31:36.012: //5384/887F0CE886A9/SIP/Msg/ccsipDisplayMsg: Sent: INVITE
*May 3 11:31:36.060: //5384/887F0CE886A9/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 407 Proxy Authentication Required
*May 3 11:31:36.068: //5384/887F0CE886A9/SIP/Msg/ccsipDisplayMsg: Sent: INVITE
The process duration is 50 milliseconds.
I suggest you to remove unnecessary sip-profile.

2. Analog Phone me simulates a PBX. How do I configure instead "+421906200200" the flaps "+421906200200 - +421906200499"?

On a single FXS analog interface you don't have this possibility. Normally a PBX is connected via ISDN BRI, PRI or E&M. In this case you must have a trunk line and the calling number is sent from the PBX. I use voice translation-rule to format the number sent from PBX. E.g. the PBX sends to cisco only the last 3 digits 200 - 499. I add a translation rule to prepend the +421906200.

3. I have to give away duplicate c = line in SDP: I configured the Necessary commands but still with double c-line.

If it doesn't cause problems you can ignore it.

4. I do not know how they work voice translation-rule 1 and voice translation-rule 2 (I just copied this from one configuration).

The command "translation-profile outgoing plus" on dial-peer 20 invokes the "voice translation-profile plus" which is composed from two rules:
translate called 1
translate redirect-called 2

The translate called 1 invokes
voice translation-rule 1
rule 1 // /+/

This rule adds the "+" to your called number: you digit "00421905012256" but in the INVITE you send +00421905012256.

The translate redirect-called 2 invokes
voice translation-rule 2
rule 1 /^$2..$$/ /+421906200\1/

This rule works on redirecting number. Actually you don't use it. The rule replaces the part of the redirecting number that starts with 2.. and with +421906200.

5. Are redundant in voice configurations some commands?

In my opinion the config is ok.

Best Regards.

View solution in original post

Daniele Giordano · ‎05-04-2014

Hi.

Questions and Answers:

1. Analog phone simulates PBX, so I have to dial the phone number in international format: 00 421 905 012 256. When I give away "rule 1 / / / + /" or I change "+" to "0" or something else, call is not realized. How do I configure rule 1 to appear correctly "sip: 00421905012256@imspp.orange.sk" but not
"sip: +00421905012256@imspp.orange.sk"?

Probably the "+" is required by orange to handle the call. So you can't remove this rule. This format is present also in incoming calls:

INVITE sip:+421906200200@192.168.49.6:5060 SIP/2.0
From: "+421905012256"<tel:+421905012256>
To: <tel:+421906200200>

Tipically + is a substitute of 00. You can eventually try this rule:
rule 1 /^00421/ / +421/

In this case your outgoing INVITE will be "sip:+421905012256@imspp.orange.sk" equal to the format of incoming call numbers.

2. Ask I Orange for a change TEL URI to a SIP URI or not?

In the last trace incoming calls are correctly handled. So is not necessary to ask anything. But if you would try just for curiosity... :-).

Regards.

View solution in original post

Daniele Giordano · ‎04-19-2014

Hi, you say "In Wireshark I see that I get the Message Router SIP 401 Unauthorized" but I don't see any response in cisco debug.
Can you add the output of "debug ccsip all" after 1 minute running?

Tipically, after a 401 response, the router must send a new SIP REGISTER with authentication parameter. But this happens only if the router got a SIP response.

In the SIP message I see a private IP address:
Via: SIP/2.0/UDP 192.168.49.6:5060;branch=z9hG4bK2D496B
Contact: <sip:+421906200200@192.168.49.6:5060>
---
sip
bind control source-interface FastEthernet0/0.765
bind media source-interface FastEthernet0/0.765

Is 192.168.49.6 the WAN IP of the router?
Or is it the LAN IP?

You must use WAN IP.

Is the router behind a NAT?

Regards.

m.hyza · ‎04-22-2014

I am glad you wrote me a message. I see RESPONSE message only in Wireshark, the router does not receive message. The IP address 192.168.4.9 is the WAN IP address, LAN IP address is not, because the router is directly connected via BRI or PRI interfase to PBX.

Regards.

Martin

m.hyza · ‎04-22-2014

Daniele,

I forgot to write that this is a test environment, therefore the use of private IP address. I do not use NAT.

Regards.

Martin

Daniele Giordano · ‎04-22-2014

Ok, the question is: why the 401 response does not reach the 2801?
What is the topology?
Cisco 2801 192.168.49.6 ---> 192.168.49.x router ----> internet
In which network segment do you have used wireshark?

Regards.

m.hyza · ‎04-22-2014

Daniele,

the topology is as follows : FXS analog . tel . - CPE: Cisco 2801 ( WAN 192.168.49.6 ) - PE: WAN: 192.168.49.5 - SBC ( 213151230248 ). Communication does not go through the Internet, but through a VPN . Wireshark is used between PE - CE routers .

Daniele Giordano · ‎04-22-2014

Can you attach the wireshark trace? We must find what block the 401.

Regards.

m.hyza · ‎04-22-2014

OK Daniele,

I attach a file with CPE configuration, debug ccsip all and Wireshark.

Regards.

Daniele Giordano · ‎04-22-2014

Please, remove sip passwords from config.

Your 2801 doesn't receive the 401. This is clear from debug ccsip output.

I don't see incoming ACL on 2801.
I don't understand who dropped this packet.
I don't understand where you get the wireshark trace.

2801 CPE----wireshark--- CE router---VPN
or
2801 CPE --- CE router ---- wireshark --- VPN

Can you check the CE router with IP address 192.168.49.5?

In wireshark trace I see different udp source ports:

Internet Protocol Version 4, Src: 192.168.49.6 (192.168.49.6), Dst: 213.151.230.248 (213.151.230.248)
User Datagram Protocol, Src Port: 60048 (60048), Dst Port: sip (5060)
Session Initiation Protocol (REGISTER)

Internet Protocol Version 4, Src: 213.151.207.112 (213.151.207.112), Dst: 213.151.230.248 (213.151.230.248)
User Datagram Protocol, Src Port: 51828 (51828), Dst Port: sip (5060)
Session Initiation Protocol (REGISTER)

...

There is something before the 2801 that blocks incoming sip message.

BR

m.hyza · ‎04-23-2014

Daniele,

thank you for your help. I managed to register CPE router - the problem was in the control-plane, that when I turned off, so sign up. But now I have a problem with INVITE messages, the CPE router does not send them.

The Call Setup Information is:
Call Control Block (CCB) : 0x68CB0F70
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : +421906200200
Called Number            : +4210905012256
Source IP Address (Sig ): 192.168.49.6
Destn SIP Req Addr:Port : :5060
Destn SIP Resp Addr:Port : :5060
Destination Name         : imspp.orange.sk

*Apr 23 11:20:14.741: //328/3A69C0708009/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 200

*Apr 23 11:20:14.741: //328/3A69C0708009/SIP/Info/sipSPIUdeleteccCallIdFromTable: Removing call id 148
*Apr 23 11:20:14.741: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[326] removed.
*Apr 23 11:20:14.741: //328/3A69C0708009/SIP/Info/sipSPIUdeleteCcbFromUACTable: ****Deleting from UAC table.
*Apr 23 11:20:14.745: //328/3A69C0708009/SIP/Info/sipSP
SIP#IUdeleteCcbFromTable: Deleting from table. ccb=0x68CB0F70 key=4547BBDB-C9FF11E3-800EE1E3-24D92598@192.168.49.6
*Apr 23 11:20:14.745: //328/3A69C0708009/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
*Apr 23 11:20:14.745: //328/3A69C0708009/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed
*Apr 23 11:20:14.745: //328/3A69C0708009/SIP/Info/ccsip_offer_ans_delete:
*Apr 23 11:20:14.745: //328/3A69C0708009/SIP/Info/ccsip_iwf_delete:
*Apr 23 11:20:14.745: //328/3A69C0708009/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 68CB0F70
*Apr 23 11:20:14.745: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[326]
SIP#
*Apr 23 11:21:19.594: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x68CB0F70) with key=[327] to table
*Apr 23 11:21:19.594: //-1/000000000000/SIP/Info/ccsip_offer_ans_init:
*Apr 23 11:21:19.594: //-1/000000000000/SIP/Info/ccsip_iwf_init:
*Apr 23 11:21:19.594: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init:
*Apr 23 11:21:19.594: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization: Entry...
*Apr 23 11:21:19.594: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetSipProfilesTag: voice class SIP Profiles tag is set : 1
*Apr 23 11:21:19.594: //-1/000000000000/SIP/Info/ccsipRegisterSetTargetInfo: p2p mode with Registrar Server = dns:imspp.orange.sk
*Apr 23 11:21:19.594: //-1/000000000000/SIP/Info/ccsipRegisterSetTargetInfo: Parsing The Registrar Address
*Apr 23 11:21:19.594: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: target_host : imspp.orange.sk target_port : 5060

*Apr 23 11:21:19.594: //-1/000000000000/SIP/Info/sipSPIValidateAndCopyOutboundHost: CCSIP: copy target_host to outbound_host
*Apr 23 11:21:19.594: //-1/000000000000/SIP/Info/sipSPIOutboundProxyReuse: Do not reuse Outbound Proxy IP adress and Port
*Apr 23 11:21:19.594: //-1/000000000000/SIP/State/sipSPIChangeState: 0x68CB0F70 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
*Apr 23 11:21:19.594: //-1/000000000000/SIP/Info/ccsip_spi_registrar_add_expires_header: Inside ccsip_spi_registrar_add_expires_header for Expires
*Apr 23 11:21:19.594: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_OUTBOUND_REGISTER
*Apr 23 11:21:19.594: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIIncrementOverloadCount: Local 1 Global 1
*Apr 23 11:21:19.594: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 40
*Apr 23 11:21:19.598: //-1/000000000000/SIP/Info/act_idle_outgoing_register: In act_idle_outgoing_register

*Apr 23 11:21:19.598: //329/000000000000/SIP/Info/act_idle_outgoing_register: Se
SIP#nd REGISTER to imspp.orange.sk:5060

*Apr 23 11:21:19.598: //329/000000000000/SIP/Info/sipSPIUaddCcbToUACTable: ****Adding to UAC table.
*Apr 23 11:21:19.598: //329/000000000000/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x68CB0F70 key=479D7871-C96111E3-8002E1E3-24D92598
*Apr 23 11:21:19.598: //329/000000000000/SIP/Info/act_idle_outgoing_register: Locally Resolved IP:213.151.230.248:5060
*Apr 23 11:21:19.598: //329/000000000000/SIP/Info/resolve_sig_ip_address_to_bind: signaling bind address : 192.168.49.6
*Apr 23 11:21:19.598: //329/000000000000/SIP/Info/sipSPIPresendProcessing: Presend Processing called for 7 event
*Apr 23 11:21:19.598: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIRetrieveOutgoingPassThruData: Retrieving Data from RCB
*Apr 23 11:21:19.598: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIRetrieveOutgoingPassThruData: Retrieving Data from RCB
*Apr 23 11:21:19.598: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone METDST to SIP default timezone = GMT
*Apr 23 11:21:19.598: //329/000000000000/SIP/Info/sipSPISendRegister: Associated container=0x68F17F24 to Register
*Apr 23 11:21:19.598: //329/000000000000/SIP/Transport/sipSPISendRegister: Sending REGISTER to the transport layer
*Apr 23 11:21:19.598: //329/000000000000/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
*Apr 23 11:21:19.598: //329/000000000000/SIP/Transport/sipSPITransportSendMessage: msg=0x691893D4, addr=213.151.230.248, port=5060, sentBy_port=0, local_addr=192.168.49.6, is_req=1, transport=1, switch=0, callBack=0x61821EB4
*Apr 23 11:21:19.598: //329/000000000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Apr 23 11:21:19.598: //329/000000000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Apr 23 11:21:19.602: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:213.151.230.248, rport:5060 with laddr:192.168.49.6

*Apr 23 11:21:19.602: //329/000000000000/SIP/Transport
SIP#/sipTransportLogicSendMsg: Set to send the msg=0x691893D4
*Apr 23 11:21:19.602: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x691893D4, addr=213.151.230.248, port=5060, local_addr=192.168.49.6, connId=3 for UDP
*Apr 23 11:21:19.602: //329/000000000000/SIP/State/sipSPIChangeState: 0x68CB0F70 : State change from (STATE_IDLE, SUBSTATE_NONE) to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
*Apr 23 11:21:19.602: //329/000000000000/SIP/State/sipSPIChangeState: 0x68CB0F70 : State change from (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE) to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)

Regards.

Martin

Daniele Giordano · ‎04-23-2014

Hi Martin.

Can you add a new debug:

debug ccsip message

debug voice ccaip inout

Regards.

m.hyza · ‎04-24-2014

Hi Daniele,

I'm sending to you debugs. I do not know if I have a well-configured dial-peer voice translation-rule.
Numbering plan shall be as follows:
Outgoing prefix = 0 (calls outside the group or "long" number)
Mobile prefix = 6 (calling on mobile phone within the group Short Number)
PBX prefix = 3 (Mobile calling within the group for a fixed flap over prefix)

PBX prefix: 906200xxx, where xxx is the range of 200-499.

Regards.

Daniele Giordano · ‎04-24-2014

Hi, dial-peers and translation-rules are ok. What is your doubt?

About outgoing INVITE, your request get 480 Temporarily unavailable from orange.
This is the number sent to orange: 0905012256.

Is right?

Can you add a full wireshark trace?

Regards.

m.hyza · ‎04-25-2014

Hi Daniele,

today I was able to call from my mobile phone 0905 012 256 to an analog telephone 0906 200 200. Analog phone rings, but I don´t hear this ringing in my mobile. But when I want to call from an analog phone to my mobile phone, so he will not allow. I hear the marriage: the unknown phone call number. I think the problem is in the voice translation-rule.

I´m sending to you CPE router configuration + debug ccsip al + wireshark trace (please rename file m.hyza_20140425.rtf to m.hyza_20140425.pcap.

Regards.

Daniele Giordano · ‎04-25-2014

Hi. Missing ringback tone during incoming calls is caused by a 183 (SDP) SIP message sent from your cisco. This is a progress message. We must send a 180 RINGING alert message.

Try to add these commands in config:

voice call send-alert
voice rtp send-recv

sip-ua
disable-early-media 180

Regarding outgoing calls, I've a doubt on the format of called number.
In wireshark traces, the called number is

Request-Line: INVITE sip:+42100905012256@imspp.orange.sk:5060 SIP/2.0

and the response from orange is a 480 Temporarily unavailable.

In the cisco debug output the called number is

INVITE sip:+0905012256@imspp.orange.sk:5060 SIP/2.0

and the response from orange is a

SIP/2.0 404  Not Found

Reason: Q.850;cause=1;text="Unallocated (unassigned) number",SIP;cause=404

Do you have tested a different translation rule?

We can try to replace + with 00 or remove international prefix.

Can you ask to orange what is the right format?

BR