Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 
Announcements

Welcome to Cisco Support Community. We would love to have your feedback.

For an introduction to the new site, click here. And see here for current known issues.

New Member

Problem with registration Cisco 2801 as SIP gateway with SIP server

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

2 ACCEPTED SOLUTIONS

Accepted Solutions

HI.Questions and Answers:1.

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.

Hi.

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.

59 REPLIES

Hi, you say "In Wireshark I

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.

New Member

I am glad you wrote me a

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

New Member

Daniele,I forgot to write

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

Ok, the question is: why the

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.

New Member

Daniele,the topology is as

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 .

Can you attach the wireshark

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

Regards.

New Member

OK Daniele,I attach a file

OK Daniele,

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

 

Regards.

Please, remove sip passwords

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

New Member

Daniele,thank you for your

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

Hi Martin.Can you add a new

Hi Martin.

Can you add a new debug:

debug ccsip message

debug voice ccaip inout

 

Regards.

New Member

Hi Daniele, I'm sending to

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.

 

Hi, dial-peers and

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.

New Member

Hi Daniele,today I was able

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.

Hi. Missing ringback tone

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

 

 

 

Another note: your outgoing

Another note: your outgoing SIP INVITE has a duplicate c= line in SDP.
Sometimes this can cause problem. See this article http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/116010-dup-c-lines-problem-solution-00.html

Regards.

New Member

Hi Daniele,thank you for your

Hi Daniele,

thank you for your advice. We have a little progress.


1.) I configured the router commands that you wrote.
2.) I realize that probably is not the right format called tel. numbers.
3.) Duplicate c = line in SDP: I configured the necessary commands but still is double c-line.
4.) I haven´t output from Wireshark, because it used my colleague.
5.) Current status is as follows:

A) Call: mobile phone (+421 905 012 256) - analog phone (+421 906 200 200):
- analog phone rings, but I can not hear your mobile phone ring tone
- I hear the call of analog phones, but it can not hear your mobile phone

B) Call: analog phone (+421 906 200 200) - mobile phone (+421 905 012 256):
- mobile phone rings and I hear in analog phone ring tone
- I hear the call in analog phones, but it can not hear in mobile phone
- is a long time since dial a phone number and hear a ring tone

Regards.

Hi.For ringback tone issue on

Hi.

For ringback tone issue on scenario A try this:

- add tone ringback alert-no-pi to dial-peer

In this case, the "183 progress" sent from cisco shold be replaced by a "180 ringing".

 

For delay after dialing try this:

- add timeouts interdigit 4 to voice-port 0/0/0

When in your dial-peers is present a destination-pattern .T, the cisco uses a default interdigit timer of 10 second to collect new digit. After this time the cisco sends the call.

 

For one way audio issue try this:

- add the route ip route 213.151.230.249 255.255.255.255 FastEthernet0/0.765 192.168.49.5 permanent

In SDP sent from orange the media ip is c=IN IP4 213.151.230.249. Please check IP connectivity.

 

Best Regards.

 

New Member

Hi Daniele,again we are a

Hi Daniele,

again we are a little closer to the target.

1.) I configured the router commands that you wrote.
2.) I realize that probably is not the right format called phone numbers.
3.) Duplicate c = line in SDP: I configured the necessary commands but still is double c-line.
4.) Please rename file m.hyza_20140429.rtf to m.hyza_20140429.pcap.
5.) Current status is as follows:

A) Call: mobile phone (+421 905 012 256) - analog phone (+421 906 200 200):
- phone is ringing and I hear the call, it is OK

B) Call: analog phone (+421 906 200 200) - mobile phone (+421 905 012 256):

- phone is ringing and I hear the call, it is OK
- is a long time (27 seconds) since dial a phone number and I hear a ring tone
- it is bad format called phone number

Best Regards.

Hi.Your incoming call from

Hi.
Your incoming call from mobile to analog phone has a problem.
There are too many retransmissions.
First of all this call uses a tel uri scheme instead of sip uri. Is it right?

I think that orange/SBC doesn't understand cisco responses.

 

Regarding the delay before the ringback tone, I've studied logs and traces.
In appearance there are about 6 - 7 seconds between end of dialing and ringing.

Can you debug the fxs port using "debug vpm signal"?

 

Regards.

 

New Member

Hi,I deleted some probably

Hi,

I deleted some probably unnecessary commands and the result is the same. Orange SBC I'll check. I can´t even rely on the abbreviated number (eg. 112, 158).
 
Regards.

Hi.Please provide me a "debug

Hi.

Please provide me a "debug vpm signal + debug ccsip message" (not all) of an outgoing call from analog to mobile. In this way I can investigate about delay before ringback.

Please previde a "debug ccsip message" (not all) of an ougoing call to abbreviate number. What is the nature of these numbers? Emercengy numers?

Ask to orange why incoming calls use TEL URI.

 

BR

New Member

Hi,I´m sending you debugs.

Hi,

I´m sending you debugs. Yes, abbreviate numbers are emergency numbers in Slovakia (112, 150, 155, 158, 159). I send a request to Orange for a change TEL URI to a SIP URI?

Best regards.

Hi. Here my considerations.

Hi. Here my considerations.

DELAY BEFORE RINGING

This is your call:

*May  2 09:14:15.325: htsp_process_event: [0/0/0, FXSLS_ONHOOK, E_DSP_SIG_1100]fxsls_onhook_offhook htsp_setup_ind
*May  2 09:14:15.325: [0/0/0] get_local_station_id calling num=+421906200200 calling name= calling time=05/02 09:14  orig called=
*May  2 09:14:15.333: htsp_process_event: [0/0/0, FXSLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]fxsls_check_auto_call 
SIP#
*May  2 09:14:17.065: htsp_digit_ready(0/0/0): digit = 0
*May  2 09:14:17.245: htsp_digit_ready(0/0/0): digit = 0
*May  2 09:14:17.713: htsp_digit_ready(0/0/0): digit = 4
*May  2 09:14:17.893: htsp_digit_ready(0/0/0): digit = 2
SIP#
*May  2 09:14:18.105: htsp_digit_ready(0/0/0): digit = 1
*May  2 09:14:18.653: htsp_digit_ready(0/0/0): digit = 9
*May  2 09:14:18.865: htsp_digit_ready(0/0/0): digit = 0
SIP#
*May  2 09:14:19.125: htsp_digit_ready(0/0/0): digit = 5
*May  2 09:14:19.453: htsp_digit_ready(0/0/0): digit = 0
*May  2 09:14:19.825: htsp_digit_ready(0/0/0): digit = 1
*May  2 09:14:20.005: htsp_digit_ready(0/0/0): digit = 2
SIP#
*May  2 09:14:20.285: htsp_digit_ready(0/0/0): digit = 2
*May  2 09:14:20.525: htsp_digit_ready(0/0/0): digit = 5
*May  2 09:14:20.793: htsp_digit_ready(0/0/0): digit = 6

YOU HAVE PRESSED THE LAST DIGIT OF CALLED NUMBER AT THIS TIME: 09:14:20.793
THE CISCO STARTS TIMEOUT INTERDIGIT OF 10 SECONDS. 

*May  2 09:14:30.797: htsp_process_event: [0/0/0, FXSLS_OFFHOOK, E_HTSP_PROCEEDING]

NOW THE CISCO TRIES A DNS SRV LOOKUP FOR ABOUT 18 SECONDS

//-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: TYPE SRV query for _sip._udp.imspp.orange.sk and type:1
SIP#
 //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_aaaa_query: DNS query for imspp.orange.sk and type:1

*May  2 09:14:48.809: //4724/33F69CFA852A/SIP/Msg/ccsipDisplayMsg:
  Sent: 
INVITE sip:+00421905012256@imspp.orange.sk:5060 SIP/2.0

 

 

So, the delay is a combination of two factors:
- timeout interdigit of 10 seconds
- dns lookup of 18 seconds

You can try these configuration:
1) "timeouts interdigit 4" to voice-port 0/0/0 to decrease first timer

2) replace "sip-server dns:imspp.orange.sk" with "sip-server ipv4:213.151.230.248"

or check dns resolving using ping imspp.orange.sk and configure ip host command to get srv record

ip host imspp.orange.sk 213.151.230.248
ip host _sip._udp.imspp.orange.sk srv 1 1 5060 imspp.orange.sk

or enable dns lookup using "ip domain lookup" and "ip name-server YOUR DNS SERVER IP"

 

 

New Member

Hi,your advice helped me

Hi,

your advice helped me again.

Changes in the configuration of CPE router:
ip host _sip._udp.imspp.orange.sk cf 1 1 5060 imspp.orange.sk     add

voice-port 0/0/0
 timeouts interdigit 3                        add

Result:
1. The delay is reduced from 28 seconds to 9.7 seconds, which is probably sufficient.
2. Call from analog phone to emergency number "00 421 2 112", where "2" is the telephone dialing code for Bratislava.

Questions:
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 ???"

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

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

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

5. Are redundant in voice configurations some commands?

Best Regards.

HI.Questions and Answers:1.

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.

New Member

Hi,I have another questions.

Hi,

I have another questions.

 

Questions:

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"?

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

Best Regards.

Hi.

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.

New Member

Hi,Questions and Answers:1.

Hi,

Questions and Answers:

1. When I configured "rule 1 /^00421/ /+421/" so I had a good format INVITE messages: To: <sip:+421905012256@imspp.orange.sk>, but I´m not called to phone number 00 421 905 012 256.


2. I configured "dial-peer voice 112 voip" and "voip dial-peer voice 150 voip" and I called to emergency numbers 112, 150, 155, 158, 159.


3. How do I configure a rule that I have not called the number in international format 00 421 905 012 256, but 00 905 012 256 or 000 421 905 012 256?

Best Regards.

 

Hi.Probably Orange supports

Hi.
Probably Orange supports the format +00 421 etc.
Do you have already tried to remove the translation rule and call only the number 905 012 256?
Should be the national format.
What will happen?

If you want call a number in national format and add a pefix you can write a simply translation rule which prepends a prefix:

!-- add the prefix +00421 to every called numbers
voice translation-rule 2
 rule 1 /^\([0-9]\)/ /+00412\1/

!-- add the prefix +00421 to every called numbers that begins with 9
voice translation-rule 3
 rule 1 /^9/ /+004129/

You can find more infos and examples here
http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/61083-voice-transla-rules.html

http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/64020-number-voice-translation-profiles.html

 

Regards.

1558
Views
108
Helpful
59
Replies