04-16-2014 12:05 AM
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
Solved! Go to Solution.
05-06-2014 08:19 AM
Hi,
I canceled the "rule 1 / / / + /" and I called the phone number 0905 012 256 and the result is as follows. Orange SBC needs format phone number +421 XXX XXX, respectively, treated with the format +00421, but this can be a mistake on the part of the SBC. I put this check in Orange.I have also configured the same rules that you wrote me, but did not help.
Best Regards.
06-09-2014 09:57 AM
Hi, try with a voice translation rule as explained in a prevoisu post.
Regards.
07-01-2014 08:40 AM
Hi jwood, in your log you can see a call destinatated to "INVITE sip:17852465184" but you don't have a dial-peer with this called number and so your CMR answer with a 403 forbidden.
Add a dp with an "incoming called-number 17852465184" and use an appropriate session-target to route the call to the CME IP address.
Regards.
07-01-2014 08:53 AM
I did some moods to the config since I posted, trying to figure it out...still now success. Do you mind reviewing my config again and see if you can point out what is wrong? Still can't dial my DID and they ring the IP phones. I am new, so if you could also help with the config layout as well I would greatly appreciate it.
If it's not too hard, can the 913 ring both phones and 785 ring just 1002?
DID's 9137129524 & 7852465184
Thank you so much for your help
07-02-2014 06:33 AM
Hi Daniele,
could you help me please? I get the following connections:
ISDN measurements IBT-300 - ISDN PRI card VWIC2-2MFT-T1/E1 - Cisco 2901 - Orange SK SBC
When I calling from the mobile number 0905 012 256 to ISDN number 0906 200 201, I dont heared the ringtone in my mobile phone, IBT-300 to ring, but when I talking to my mobile phone, so I nothing heard in IBT-300. The same is true when I call from IBT-300 to mobile phone. You know you help me?
Best Regards.
05-07-2014 09:00 AM
Hi,
I have some questions and answers:
1. +00 was a mistake on the part of Orange SBC. Now I already call 0 00 421 905 012 256 or 0 0905 012 256. It´s OK.
2. I joined second phone and all calls to second phone are OK, just can not call from the second phone. I do not know whether the problem is in the other phone registrations
3. I could reduce delay before ringing:
analog phone - mobile phone = 10 seconds NOK
mobile phone - analog phone = 6 seconds OK
Best Regards.
05-07-2014 11:05 AM
Hi.
All outgoing calls from the "second phone" are rejected by orange with a "SIP/2.0 403 Forbidden".
This means that the outgoing calls from this number are blocked from orange.
In the "show run" I don't see a specific configuration for this number.
Is this an alias (a secondary number) of the +421906200200?
In this case the orange forbidden could be right.
Or do you have a different authentication credential for this second number?
If yes you must configure it under sip-ua menu.
E.g.
sip-ua
credentials username 0000095901 password *** realm voice.ae.net
credentials username 0000095902 password *** realm voice.ae.net
credentials username 0000095903 password *** realm voice.ae.net
credentials username 000009591 password *** realm voice.ae.net
authentication username 000009591 password 7 *** realm voice.ae.net
disable-early-media 180
retry invite 3
timers expires 60000
registrar dns:voice.ae.net expires 300
sip-server dns:voice.ae.net
no suspend-resume
You can also ask to orange for more detail.
About the delay before ringing you wrote:
analog phone - mobile phone = 10 seconds NOK
6 or 7 second are the normal call setup over mobile network.
The other 3 second are the timeout interdigit delay of cisco.
This delay is due to the dialing methods using in SIP: en-block.
The gateway collects digits and sent a single block.
The traditional PSTN/TDM network uses the overlap dialing. In this case each digit is sent immediatly to the switch and the setup time is very short.
In Italy we have a variable lenght dial-plan and so we must use en-block with interdigit delay. The en-block timer is handled from T in dial-peer:
destination-pattern [0,6,3]T
Is the same in your country?
If in your country there are a fixed lenght dial-plan (like in US where the number has a lenght of 10 digit ) you can modify the dial-peer removing T
destination-pattern [0,6,3].........
In this case the number is sent immediatly without delay.
To reduce the setup time you can also dial the # at the end of the number. This inform the cisco that the number is complete and must be sent immediatly.
E.g. 000421905012256#
Regards.
05-12-2014 07:49 AM
Hi,
1. I added second analog phone (voice-port 0/0/1, dial-peer voice 201 pots, +421906200201). I'm also reachable from 201 and to 201. Configuration was changed from: P-Asserted-Identity: <sip:+421906200201@imspp.orange.sk> to P-Asserted-Identity: <sip:+421906200200@imspp.orange.sk>. It´s OK?
2. Now I'm going to address faxes.
3. May I change P-Asserted-Identity in message BYE from <sip:+421906200201@imspp.orange.sk> to <sip:+421906200200@imspp.orange.sk>?
Best Regards.
05-12-2014 09:41 AM
Hi, the config is OK. If you want modify the P-Asserted-ID in the BYE request you must add a new rule in your SIP PROFILE. You can build a rule for different request methods:
router(config-class)#request ? ACK sip ack ANY any sip request BYE sip bye CANCEL sip cancel COMET sip comet INFO sip info INVITE sip invite NOTIFY sip notify OPTIONS sip options PRACK sip prack PUBLISH sip publish REFER sip refer REGISTER sip register REINVITE sip reinvite SUBSCRIBE sip subscribe UPDATE sip info
If you want, you can use the ANY statement to modify the PAI in every requests:
request ANY
Regards.
05-13-2014 08:25 AM
Hi,
I have the following problems now:
1. (D) +421906200201 also registers, what is wrong, because to be registered only +421906200200. How do I remove it?
2. (C) Between registration +421906200200 and additional registration +421906200200 is a certain time (about 15 seconds), if the router is not registered. Is it right or wrong?
3. (C) Which means P-Associ-URI: +42190620020, +421906200! [, +421906200! [, +42190620020. This is probably wrong, how do I fix it?
4. (A, B) I joined 2 faxes:
- fax PSTN (type: Panasonic KX-FP701, +421 2 5341 82 73)
- fax router (model KX-FL403, +421906200201, voice-port 0/0/1)
From fax PSTN I sent fax message to router, but from fax router I don´t sent fax message to the fax PSTN. Why? Wireshark is the following message: (T.38: Malformed Packet: T.38).
5. Please rename files from rtf to pcap.
Best Regards.
05-13-2014 10:53 AM
Hi.
QUESTIONS and ANSWERS
1. (D) +421906200201 also registers, what is wrong, because to be registered only +421906200200. How do I remove it?
Add the command "no sip-register" under dial-peer 201 pots. Check the registration using "show sip-ua register status" command.
2. (C) Between registration +421906200200 and additional registration +421906200200 is a certain time (about 15 seconds), if the router is not registered. Is it right or wrong?
You can check the registration timer using "show sip-ua register status" command.
The first registration has a little delay but every re-registration process starts always before the expiration time. In every case, the registration timeout is negotiated during a SIP REGISTRATION in REGISTER-200 OK messages and the router uses always the lowest value. In your case the SBC forces the timer to "expires=220" so 220 seconds. This behaviour is normal.
3. (C) Which means P-Associ-URI: +42190620020, +421906200! [, +421906200! [, +42190620020. This is probably wrong, how do I fix it?
This is a fresh IOS feature. There are no problem. You can ignore it. More info here: http://www.cisco.com/c/en/us/td/docs/ios/voice/cube/configuration/guide/vb_book/vb_book/vb_9397.html
P-Associated URI
The Cisco Unified Border Element supports the use of PAURI headers sent as part of the registration process. After the Cisco Unified Border Element sends REGISTER messages using the configured E.164 number, it receives a 200 OK message with one or more PAURIs. The number in the first PAURI (if present) must match the contract number. The Cisco Unified Border Element supports a maximum of six PAURIs for each registration.
Note The Cisco Unified Border Element performs the validation process only when a PAURI is present in the 200 OK response.
The registration validation process works as follows:
•The Cisco Unified Border Element receives a REGISTER response message that includes PAURI headers that include the contract number and up to five secondary numbers.
•The Cisco Unified Border Element validates the contract number against the E.164 number that it is registering:
–If the values match, the Cisco Unified Border Element completes the registration process and stores the PAURI value. This allows administration tools to view or retrieve the PAURI if needed.
-If the values do not match, the Cisco Unified Border Element unregisters and then reregisters the contract number. The Cisco Unified Border Element performs this step until the values match.
4. (A, B) I joined 2 faxes:
- fax PSTN (type: Panasonic KX-FP701, +421 2 5341 82 73)
- fax router (model KX-FL403, +421906200201, voice-port 0/0/1)
From fax PSTN I sent fax message to router, but from fax router I don´t sent fax message to the fax PSTN. Why? Wireshark is the following message: (T.38: Malformed Packet: T.38).
There is a codec negotiation error.
In this case your cisco detects a fax and it tries to switch to T.38 fax relay but in SIP/SDP there is alto a vocal codec. There are three connection header:
After that, it tries to negotiate T.38 only but there are too many information:
Could be an IOS problem. Can you upgrade IOS and try again?
Regards.
05-14-2014 10:55 AM
Hi,
1. I configured on dial-peer voice 201 pots "no sip-register" and it is not registered +421906200201, it is OK.
2. I changed the configuration, but the fax problem persists. I do not know whether it is a good idea to change the IOS. I would like to
the issue is resolved otherwise. I also noticed that there 3x connection informations.
3. Please rename file from rtf to pcap.
Best Regards.
05-14-2014 10:55 AM
Hi, the 403 Forbidden is provided by orange because in your outgoing INVITE the P-Asserted ID is missing. Please add again
voice service voip sip asserted-id pai
The outgoing fax issue is caused by interoperability problems with T.38.
Can you try to use G.711 fax pass-through insted of T.38. It's less reliable but more easy and interoperable.
To disable T.38 and enable fax passthrough use these commands:
voice service voip
no fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
fax protocol pass-through g711alaw
modem passthrough nse codec g711alaw
dail-peer voice ...
fax rate disable
fax protocol pass-through g711alaw
If you want use T.38 try to remove duplicate C= line in SDP using these commands:
voice class sip-profiles 1
request INVITE sip-header P-Asserted-Identity modify "(P-Asserted-Identity:) (.*<sip:)(.*)(@.*)" "\1<sip:+421906200200@imspp.orange.sk>"
request INVITE sip-header P-Asserted-Identity modify "(P-Asserted-Identity: <sip:)(.*)(@.*)" "\1+421906200200@imspp.orange.sk>"
request REINVITE sdp-header Connection-Info remove
response 200 sdp-header Connection-Info remove
Regards.
05-15-2014 08:21 AM
Hi,
I changed configuration of the fax from T.38 to G.711 and faxs at work OK. I deleted dial-peer voice 21, 112, 150 and I replaced them with voice-translation-rule 1.
Questions:
1. I need to configure CLIR. A colleague told me that Orange SBC requires that I sent from router %2331%23 and therefore I should configure:
rule 6 /^#31#/ /%2331%23/
voice class sip-profiles 1
request INVITE sip-header Privacy add "Privacy: id"
but the router not take /%2331%23/. Where is the problem?
2. I can´t call to abbreviated numbers (T-com info links), for example 0 1181 and 0 12 111.
Best Regards.
05-15-2014 09:41 AM
Hi.
#31# is an ETSI (European Telecommunications Standard Institute) service code to set "per call clir".
The # is automatically translated to %23 in sip signalling.
You simply dial #31# before the called number.
About T-com info links, I'm sorry but I don't know these numbers. Your call are rejected with 480 Temporarily unavailable. Probably the number that you have dialed is not correct or is not reachable. Ask to Orange for more details.
Regards.
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