04-05-2007 12:04 AM - edited 03-14-2019 08:51 PM
I have sip account from provider and config to sip-ua with cisco 3800 series all peer behind my pbx are registered then I have call to some telephone number
I have hear from IVR of sip server "this's time number is not valid".
What's the "time number" that the sip server want? what command can solve this problem?
!
dial-peer voice 3 voip
destination-pattern T
redirect ip2ip
voice-class codec 1
voice-class sip transport switch udp tcp
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
acc-qos guaranteed-delay audio
!
!
sip-ua
authentication username **** password ****
no remote-party-id
retry invite 3
retry response 3
retry bye 3
retry cancel 3
retry register 10
timers connect 100
timers connection aging 30
mwi-server ***ip*** expires 3600 port 5060 transport udp unsolicited
registrar ***ip*** expires 3600
sip-server ***ip***
notify telephone-event max-duration 3000
!
Thank you.
04-21-2007 05:10 AM
Yes, I had forgot that you told us the strange message about "time".
Now if you want to try to change silence suppression configure "no vad" under dial-peer and see if that helps.
04-21-2007 06:24 AM
Hi,
I was looking again at the trace. The call appears to be made to 11 digits number. I understand that Thailand uses 7 digits plus area code, else if dialing internationally you may need to prefix with 00.
04-23-2007 02:37 AM
Now we have a bit more explaination.
Caller Send INVITE
SER/SIP Proxy send 100
SIPPY (an IVR, I believe is *) send 183 Session Progress, along with early media (one-way audio) to announce to Caller that 'something is invalid', last about 6 seconds.
Then the caller hung up.
That's explain the time taken between 183 and CANCEL.
Now, let see if you sip-ua successfully registered with your provider,
Pls issue this command : sho sip-ua register status
Your debug trace shows that the caller still be allowed to send INVITE regardless the sip-ua register status.
To find out what actually happens to your original INVITE, debug ccsip message, ask the caller to hold the line even after the IVR message, I expect something like 4XX respone.
To solve issue, you need help from your provider to inform you what is not ok from your side.
Thanks
SS
04-23-2007 02:57 AM
Hi ngss,
as I was mentioning before, if the called number is actually the one present in initial trace, it doesn't make sense, as the ITSP appears to based in Thailand.
04-23-2007 04:18 AM
Yes, some time users call to thailand number and international number, voice gw in thailand this sip server in singapore. till now they can't call to any destination and i'm tested "no vad" nothing difference. thank you so much.
04-23-2007 04:58 AM
Hi phokiszar, the thing is that being the sip server in singapore, you need to send all calls with 00 before the e.164 number, possibly only calls to singapore can be sent as national calls, but you should check this with the ITSP.
Please use an translation-profile to add 00 or tell you users to call with 00...
If you want to catch calls to Thailand and then add 00 and CC this is also possible, again using the translation-profiles.
Hope this helps, if so please rate post!
05-10-2007 07:57 PM
I have tested the translation rule is the same. I'm talking with provider they give me some information, log from server just like this.
Calling-Station-Id = 'None'
May 11 11:45:48: Authorization failed: Failed - Invalid Account number
May 11 11:45:48: Authentication reject response
from the log above "Calling-Station-Id = 'None'" which command or parameter can make this field have calling-id?
thank you all.
05-11-2007 03:18 AM
Hello,
from the log, it seems that you are using "200' as username. However, the ITSP never challenges for authentication.
Is this "200" the username that the ITSP has given you? What have you configured as "authentication" under sip-ua ?
05-14-2007 07:41 PM
Yes, the first time I use that number of dial-peer. now I have change for a while to sip number and tested then got log same above.
05-15-2007 03:44 AM
What I'm asking, is that the ISP should have give you username and password and possibly a realm, do you have configured that under sip-ua ?
05-15-2007 03:56 AM
Now i'm done this case sip provider they optimize some thing in their sip server. Maybe authentication method.
Thank you so much p.bevilacqua and other.
01-21-2008 02:17 AM
Hi,
I've noted your specific competence into Unified Communications and sip configurations so I wish to post you a question.
I've to implement multiple sip registration with a sip provider using a voice gateway; I know that is accepted only 1 authentication for router.
How can I do ?
01-21-2008 02:35 AM
Unfortunately nothing . Submit your request to Cisco for future implementation. One possible contact is Tony Huynh <tonhuynh@cisco.com>, he is CME's TME.
01-21-2008 02:42 AM
Sip provider give me 5 accounts related to 5 Pstn numbers assigned to my profile.
Now I'm able to use only 1 number (the number specified into authentication username ..)
How can I use the others ?
I've also a problem with Dtmf on sip connections ..
05-23-2007 01:47 PM
Hi Ph0kiszar,
I have an issue with my sip trunk. I'm using a CCME on 2800 router trying to register it with my ITSP using a SIP Trunk.
My configuration is:
!
dial-peer voice 800 voip
translation-profile outgoing strip-sip
destination-pattern 7[2-9]..[2-9]......
redirect ip2ip
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
acc-qos guaranteed-delay audio
!
!
!
sip-ua
authentication username xxx password xxx realm dns:sip.x.ca
no remote-party-id
retry invite 5
retry response 3
retry bye 5
retry cancel 5
retry prack 5
retry notify 4
retry register 5
retry options 5
timers connect 100
timers connection aging 30
timers register 600
registrar dns:nat.babytel.ca:5065 expires 3600
sip-server dns:sip.babytel.ca:5060
notify telephone-event max-duration 3000
!
and the outputs of the "debug ccsip messages" is:
Mar 23 17:40:38.386: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:nat.babytel.ca:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.240:5060;branch=z9hG4bK2BF6E1
From: <>14168486814@nat.babytel.ca>;tag=10A1A5C-1211>
To: <>14168486814@nat.babytel.ca>>
Date: Fri, 23 Mar 2007 17:40:38 gmt
Call-ID: 304915F7-D89B11DB-836EE10D-B064E7F7
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1174671638
CSeq: 5 REGISTER
Contact: <14168486814>14168486814>
Expires: 3600
Content-Length: 0
Mar 23 17:40:38.418: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden (Outbound Proxy Policy)
To: <>14168486814@nat.babytel.ca>;tag=6bc3de4f>
From: <>14168486814@nat.babytel.ca>;tag=10A1A5C-1211>
Via: SIP/2.0/UDP 192.168.5.240:5060;branch=z9hG4bK2BF6E1
Call-ID: 304915F7-D89B11DB-836EE10D-B064E7F7
CSeq: 5 REGISTER
Server: DITC-PeerPoint C100/3-05-26-GA7p2
Content-Length: 0
based on your experience with sip trunk can you give me a hand to solv this problem please.
I would appreciate so much your help.
Thak you!
Adrian
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