SPA9000 / WIP330 SIP setup
Hi,
I have a SPA9000 at 10.10.1.5 on a network. Setting up SPA phones was not to complex.
My problem is with the WIP330. What should be the SIP Settings of the WIP330 in this setup? Is there any type of guide on setting up a WIP phone with the SPA9000 ?
Thank you all
Mario
Took me about a minute reading through the WIP330 manual to find it (makes one wonder doesn't it.. I have no such device myself).
On page 60 of the user guide you have the SIP settings.
proxy = ip-address-spa9k
proxy port: 6060
register proxy ip: ip-address-spa9k
register proxy port: 6060
outbound proxy ip:
outbound proxy port:
expire time: 300
phone number: 1234
username: 1234
password:
That should do it (when I didn't write anything.. leave it at its default). outbound proxy ip must not contain anything.. if it does, remove it.
If you can register but not make or take any calls, then set
outbound proxy ip: ip-address-spa9k
outbound proxy port: 6060
There are two essential pieces to tracking down a problem with your VoIP equipment:
The configuration of every device involved
SIP protocol traces
And don't forget: there's no such thing as giving too much information when describing a problem.
Similar Messages
-
Bbandtalk sip setup cant make outbound calls
hi i have extracted the sip settings for my bbandtalk service and loaded them into my 3cx server and it has registered fine. However, I cant make outbound calls. It comes up with a message saying 'sorry you cannot make or receive calls on this line'? any ideas? i have seen something about you cant use the bbandtalk sip setup unless you are on your bt broadband line....but if this was so, then how does the bbandtalk softphone client work?
Having similar problems ("Sorry, you cannot make or receive calls ..."). Using FreePBX and Asterisk and could use some help with the settings. Happy to take settings that go into other configuration files, but I'm currently stuck. Here's what I've got thus far (and it doesn't work -- tried several permutations, too):
type=peer
context=incoming_bbt
username=445603449781
authuser=445603449781
secret=B9R0WDKP9KAQMVX
host=62.239.169.148
insecure=very
fromuser=445603449781
fromdomain=62.239.169.148
dtmfmode=rfc2833
Any help would be appreciated. Have other VOIP services working and BT seems to register fine. -
I just purchased a Nokia E66 and I am trying to do the SIP setup in order to use Skype. Does anyone know what to put as the Proxy Server and Realm? I am living in Qatar if that makes any difference.
Thanks for any and all help.Skype doesn't use SIP. It uses its own protocol. You won't get Skype working by doing anything in your SIP settings.
There's probably an S60 application released by Skype that allows you to use this protocol on an E66. Alternatively, I know that fring allows you to use Skype protocol.
Was this post helpful? If so, please click on the white "Kudos!" star below. Thank you! -
Linksys SPA9000 and SIP registration with an E61
HI, I have my own IP telephony but I want to register my phone to be used with my Linksys SPA9000, anybody has the setup to do this with an E61?
Unfortunately my Nokia, an N95, can't register with the SPA9000 either. I managed to 'partially' register the phone. The phone registered itself an i saw it in the pbx status page.
(http:// - ip-adres of SPA9000 - /admin/status). The phone itself keeps notifying that it couldn't establish a connection and won't connect to the PBX.
What did the trick is realising that the PBX proxy listens to port 6060. So i added that port in the phone for the proxy and registrar. I know it's a part of the solution and i am working on it.
My plan
- setting up a syslog service on my linux server and search the logs for error mesages during the registration proces
- upgrading my firmware. The SPA9000 is up-to-date, but my N95 doesn't have the latest version yet. So i want to check the release notes from Nokia for information about solved SIP errors.
- checking the forums for tips
- ask Nokia for help -
SPA9000/SPA400 Initial Setup Problems
Hi All
I hope someone out there will be able to help a newbie to VOIP as we're getting nowhere slowly and painfully with the documentation. Our setup is:
SPA9000 connected to SRW208P switch via the Internet port on the SPA9000
SPA400 connected to the SPA9000 via the switch above
SPA400 connected to a BT (British Telecom) POTS socket on port #1.
2 x Linksys SPA922 phones connected to the switch
1 x Ordinary analog phone connected to FXO Port #1 on the SPA9000
At the moment we've got no SIP service connected and hope to use the system entirely on POTS for now
So far we've run the wizard and sucessfully connected everything together, assigned IP addresses etc. The SPA9000 can "see" the SPA400 and we've triple checked the addresses, users and ports match up OK. Likewise the phones are connected and we've been able to configure them and we can make internal calls on the system OK. During the wizard configuration we've set Line #1 on the SPA9000 to the attached SPA400 and assigned 9 to choose that line.
The problems start when we try to make or receive a call on the external line via the SPA400. When trying to make a call (either dialling the number directly or prefixing with 9) then we get an engaged type sound and it doesn't ring out. On the SPA400 management page all four lines are showing as on-hook although line 1 is showing a voltage etc. The line 1 status light is also flashing very quickly at all times. The BT line is known to work and has nothing else connected to it. For incoming calls the system doesn't pick up the line and it just keeps ringing.
Finally the analog phone connected to the FXO #1 port on the SPA9000 appears completely dead with no dial tone and you can't ring out internally or externally using this phone.
We've tried numerous combinations of configuration but nothing seems to make any difference. Some advice would be very appreciated if anyone can spare the time for a quick how to guide.
Many Thanks
Andy.Andy,
I've had the same issue with a SPA9000/SPA400 install on a BT line in the UK. The FXO port LED would blink continuously and the line would appear dead - no pickup on incoming calls, and outbound calls would show service not available on the phone.
After a *lot* of experimentation, I managed to get the line stable enough to get this working, but its a workaround and not a solution.
If you syslog'ed the SPA400 you got entries such as:
Sep 14 21:51:44 192.168.0.14 0000715174 - TIU: 3 FXO voltage changed, state on, voltage 7V -> 0V^M
Sep 14 21:51:46 192.168.0.14 0000716210 - TIU: 3 FXO voltage changed, state na, voltage 0V -> 24V^M
Sep 14 21:51:47 192.168.0.14 0000716805 - TIU: 3 FXO voltage changed, state on, voltage 24V -> -5V^M
Sep 14 21:51:48 192.168.0.14 0000717304 - TIU: 3 FXO voltage changed, state on, voltage -5V -> 8V^M
Sep 14 21:51:49 192.168.0.14 0000717805 - TIU: 3 FXO voltage changed, state on, voltage 8V -> 14V^M
Sep 14 21:51:50 192.168.0.14 0000718305 - TIU: 3 FXO voltage changed, state on, voltage 14V -> 20V^M
Sep 14 21:51:51 192.168.0.14 0000718804 - TIU: 3 FXO voltage changed, state on, voltage 20V -> 23V^M
and you can see the line voltage changing continuously.
So, I added a splitter to the line at the master socket, and put a known working standard phone on the same line as the SPA400. So, this provided a stable impedance to the line, and the line settled at -47V and no voltage fluctuations.
Given that I am using 320 Ohms + (1050Ohms || 230nF) which is Linksys's suggested value, and I have tried 370 Ohms + (620 Ohms || 310nF) as shown in this BT Document (http://www.sinet.bt.com/351v4p5.pdf), and the default of 600Ohms all of which resulted in an unstable line im confused as to what BT are doing to the line.
Andy, does the incoming PSTN line you are using have ADSL on it? or did you manage to find a real solution? -
So far I have a 2811 setup with a few CIPC softphones. This is working fine.
What i'd like to do now is set up calling to external phones. We have set up a voip account with a company and have the following details. SIP username, SIP Password and SIP Proxy.
How can i go about using this to connect to an external number? The reason i'm asking for online documentation and help is that the company said they'd assist in setting it up , but they've been a little slow so far.
Many Thanks.Sorry I should have been more thourough.
Software Version 12.3(11)T10 / CME 3.2 -
Nokai E75 VOIP/SIP Setup for WLAN: sipgate.de - ...
Had problems setting up the VOIP Client on my E75 for the SIP Provider sipgate.de: Test calls to their 10005 or 10000 number worked without problems, but for outgoing calls to normal landline numbers no audio stream was connected through (ringback was ok):
Setup:
- sipgate.de configured via MENU -> CTRL PANEL -> NET SETTINGS -> Download
- SIP Profile (user/password etc. )configured based on customized config from sipgate.com
- WLAN (linksys or AVM-Fritzbox)
- Provider Tcom/Avego or 1&1
After trying lot of different things & settings it finally turned out that deleting the NAT FW settings fixed the issue:
- MENU -> CTRL PANEL -> NET SETTINGS -> ADVANCED VOIP SETTINGS
- NAT FIREWALL SETTINGS -> DOMAIN PARAMETERS
-> select sipgate.de
-> OPTIONS -> DELETE PARAMETERS
if you don't have the "Advanced VOIP Settings" Menu, you need to load a special sip addon application. Documentation and application is available at
forum.nokia.com
Search for "SIP VOIP SETTINGS"
If the above tweak also works for other WLAN Router or even UMTS Settings, I don't know.
With regards to sipgate.de --> havn't used it yet a lot, but the good thing is it's not a standalone client, but as it's reusing the Nokia VOIP Client it nicely integrates into the phone (same contact list, button to choose internet vs. mobile dialing, ..
hope the above saves some people some time
joNative VoIP/SIP is the main reason I was so pleased to switch to the E72 (I use voiptalk.org myself).
Until just over a year ago I had an N95, which also had built-in VoIP. I lost that feature when I "upgraded" to an N96 and then a 5800XM, but am pleased as Punch to have it again with the E72. It makes life so much easier having the same device acting as both mobile and "landline" (my entire phone setup at home is VoIP-based with Asterisk).
Was this post helpful? If so, please click on the white "Kudos!" star below. Thank you! -
SIP setup and the 'sip phone' symbol on E71
Hey....
OK SIP users..this question is for you.
Iam running my E71 and have activated and paid for a SIP account (voiptalk) and the setup went well last week, and I have made a few successful SIP calls from the E71.... good stuff !
I am calling landline numbers in the UK using SIP (VoipTalk) through FRING... no problems...
My question is... the little SIP-Phone symbol that was present on the top left of the screen has now disappeared... but I can make SIP calls...
Erm... why has the symbol gone ?
What does that symbol mean in reality, as I can continue to make SIP calls...
Thanks in advance
HarryOkay thats a question more to the phone's behavior rather then SIP settings.
Looks like your phone keep dropping out symbols.
Have you tried to reboot your phone lately by swithing it off removing the battery and put it back together after a couple of minutes.
Sometimes that solves a lot of issues.
Try it and see if it brings back the symbol.
Mark me a KUDOS if this has helped... -
SIP setup with Media5-fone and CUCM7
I think I have configured it all correctly and I have viewed the netcraftsmen website but it differs a little.
When I run a sip trace from Media-fone5 i see an error : Warning: 399 (CUCM name) " Line not Configured"
I can not find out what line needs to be configured.. Any ideas?Gene,
Actually, I just loaded Media5 on my iPod Touch for testing so we can add it to our blog. I haven't finished testing, but I am able to get the phone to register. A full blog is forthcoming but you may want to try this in the interim. Note, we are assuming the same basic approach as you see for the various Droid and iPhone clients Hailey and I write about in the blog.
Under the account settings main page:
Username: set this to the line DN you have assigned to the SIP device in CUCM (e.g. 7035551234)
Password: set this to the digest you assigned to the end user you created in CUCM (e.g. C1$co12345)
Servers config page:
Put the IP address of the server you want to register to. Keep the port info the same.
Advanced config page:
Auth Name: set this to the user ID assigned to the end user you created in CUCM (e.g. mysipuser)
DTMF Method: You should be fine with RFC 2833 (default) but I have not tested
Again, I just loaded this yesterday and only tested registration. I plan on going back to the config once I get some free cycles. So, we will have a write up on the blog sometime in the next week.
Let us know if the cursory overview above gets you pass your current issue.
HTH.
Regards,
Bill -
Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP
Hi Cisco Community,
I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
Below is an example of a call that is connected with the current setup:
Note:
IP: 10.18.81.2 (CUBE)
IP: 10.18.81.11 (CUCM SUB)
IP: 10.111.111.254 (ITSP SBC)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:29.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
Session-Expires: 1800
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1417347869
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 301
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
s=SIP Call
c=I
PM-HO-VG-01#N IP4 10.18.81.2
t=0 0
m=audio 22256 RTP/AVP 18 0 8 101
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf9
PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,application/xml
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC81D00
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Type: application/sdp
Content-Length: 236
v=0
o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.80.40
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
PM-HO-VG-01#
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
PM-HO-VG-01#sh sip
PM-HO-VG-01#sh sip-ua call
PM-HO-VG-01#sh sip-ua calls
Total SIP call legs:2, User Agent Client:1, User Agent Server:1
SIP UAC CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 27218091323
Called Number : 0862000000
Bit Flags : 0xC04018 0x10000100 0x0
CC Call ID : 64511
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.111.111.254]:5060
Destn SIP Resp Addr:Port: [10.111.111.254]:5060
Destination Name : 10.111.111.254
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64511
Stream Type : voice+dtmf (0)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22256
Media Dest IP Addr:Port : [10.111.111.254]:20074
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 0218091323
Called Number : 0862000000
Bit Flags : 0xC0401E 0x10000100 0x80004
CC Call ID : 64510
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.18.81.11]:5060
Destn SIP Resp Addr:Port: [10.18.81.11]:5060
Destination Name : 10.18.81.11
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64510
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22350
Media Dest IP Addr:Port : [10.18.80.40]:21928
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Server(UAS) calls: 1
PM-HO-VG-01#
PM-HO-VG-01#
PM-HO-VG-01#
As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22256 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 102 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 360
v=0
o=BroadWorks 316169737 2 IN IP4 10.111.111.254
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
a=inactive
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22350 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Length: 0
Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 103 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 306
v=0
o=BroadWorks 316169737 3 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 2
PM-HO-VG-01#00 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 213
v=0
o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.81.10
t=0 0
m=audio 4000 RTP/AVP 18
a=X-cisco-media:umoh
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=sendonly
Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 101 BYE
Reason: Q.850;cause=86
P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 104 BYE
Reason: Q.850;cause=65
P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 Race Condition
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
Timestamp: 1417347889
CSeq: 104 BYE
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 200
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 101 BYE
Content-Length: 0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 86
Disconnect Cause (SIP) : 200
PM-HO-VG-01#Hi Manish,
Again, excellent feedback. Much appreciated.
I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
I will be doing some intensive test again later on this week and will send the logs.
Here is my question to both of you:
Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
Thanks again for your support fellows. -
I have a new WRT610N. I also have a new SIP phone device for my PC. If I do not have my router in the path, I can call in and out successfully. When I put the router in the path, if I call out, no problems however when a call comes in, the call drops in 30 seconds. Using Wireshark, I have determined that the router is not allowing a SIP ACK back to the PC which is required to complete the SIP setup. All other SIP packets (Invite for example) make it through. Only the ACK is blocked. I am aware of a firewall feature that blocks SIP ACKs to prevent a denial of service attack against a SIP server, but I am only trying to pass a single SIP ACK to my phone device. Anyone have any ideas how to fix this?
Had a similar problem with my VoIP system. I first had a WRT54GC router and a VoIP phone behind that, worked without problems. Upgraded to a WRT610N router and my VoIP phone wouldn't register anymore.
Reading on here the hint that it could be a problem with the SIP ACK being blocked I researched a bit and found out that the router has a function called "Port Triggering" that is exactly designed to deal with that. It will monitor outgoing connections and allow incoming connections from same IP address.
So all I did is go to "Applications and Gaming" -> "Port Range Triggering" and put a new entry as follows:
SIP Trigger Range: 5060 - 5060 Forward Range: 5060-5060
Enabled it and the problem was solved, my VoIP phone works without a problem! -
All Day Event 2 weeks long stops mid-way through (in Week View)
I created a new all-day event that spans 2 weeks: July 6 - July 19. In Month View, it shows as a continuous line covering the 2 weeks. But in Week View, the event stops after the 10th day, on July 15...then shows up again when I click to the next week.
I tried deleting and recreating. That didn't work. I also tried changing the dates, and when it was less that 10 days, it showed up as a continuous line. But no matter what, on the 10th day, it showed a stop...then picked up again the next week.
Any suggestions?
Thanks!Hi,
interesting story. Did you find a way to configure phone for Internet calls (VOIP)? Using Setup/Connection/Admin Setup/SIP Setup (I am not sure about names as I am using Polish version) I may configure SIP profile and even register it ... but that is all. No way to make Internet calls even manual is saying about way of making them
regards, -
404 not found when trying to make outbound calls
Hi.
Just got skype for business, incoming calls working fine but cannot get outbound to work.
Any ideas?
Call-ID: YzUwYWZiNDRlMTc0NzhlNDlmYjExMmRjNWFmZDE3M2Q.
CSeq: 1 INVITE
Session-Expires: 1800
Accept: application/sdp, message/sipfrag
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, NOTIFY, INFO, REFER, PRACK
Content-Type: application/sdp
Supported: replaces, histinfo, 100rel, timer
User-Agent: TeleVoice Trio Enterprise 3.2 TE 3.2.79
History-Info: <s
v=0
o=- ****************330 1 IN IP4 192.168.0.191
s=-
c=IN IP4
t=0 0
m=audio 53018 RTP/AVP 8 101 0
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:0 pcmu/8000
sigcomp id=
Resiprocate: (level=7)
Adding message to tx buffer to: [ V4 UDP target domain= mFlowKey=0 ]
Resiprocate: (level=7)
incoming from: [ V4 UDP target domain=unspecified mFlowKey=352 ]
Resiprocate: (level=
CSeq: 1 INVITE
Content-Length: 0
Resiprocate: (level=7)
incoming from: [ V4 UDP target domain=unspecified mFlowKey=352 ]
Resiprocate: (level=
CSeq: 1 INVITE
Content-Length: 0
[edited for privacy]can anyone over at Skype help, there seems to be no online chat support anymore, and I am a paying customer, very dissapointing!
I have Skype Connect (skype sip) setup for incoming calls and this has been working well for some time, but now i want to call out to skype users using the same account and I get both 407 and 404 errors, a redacted trace copied below:
|Time | xx.xxx.xx.xxx |
| | | 63.209.144.201 |
|0.000 | INVITE SDP (g711U g711A telephone-eventRTPType...1 CN) |SIP From: "" <sip:[email protected] To:<sip:[email protected]
| |(5080) ------------------> (5060) |
|0.041 | 100 Trying| |SIP Status
| |(5080) <------------------ (5060) |
|0.184 | 407 Proxy Authentication Required |SIP Status
| |(5080) <------------------ (5060) |
|0.184 | ACK | |SIP Request
| |(5080) ------------------> (5060) |
|0.185 | INVITE SDP (g711U g711A telephone-eventRTPType...1 CN) |SIP From: "" <sip:[email protected] To:<sip:[email protected]
| |(5080) ------------------> (5060) |
|0.225 | 100 Trying| |SIP Status
| |(5080) <------------------ (5060) |
|0.857 | 404 Not Found |SIP Status
| |(5080) <------------------ (5060) |
|0.857 | ACK | |SIP Request
| |(5080) ------------------> (5060) |
the (redacted) gateway config in freeswitch is:
<include>
<gateway name="skype">
<param name="username" value="9xxxxxxxxxxxxx"/>
<param name="realm" value="sip.skype.com"/>
<param name="from-user" value="9xxxxxxxxxxxxx"/>
<param name="from-domain" value="sip.skype.com"/>
<param name="password" value="XXXXXXXXXXXXXX"/>
<param name="extension" value="9xxxxxxxxxxxxx"/>
<param name="proxy" value="sip.skype.com"/>
<!-- <param name="register-proxy" value="stun.skype.com"/> -->
<param name="register" value="true"/>
<!-- which transport to use for register -->
<param name="register-transport" value="udp"/>
<!--How many seconds before a retry when a failure or timeout occurs -->
<param name="retry-seconds" value="30"/>
<param name="caller-id-in-from" value="false"/>
<param name="contact-params" value="[email protected]"/>
<param name="context" value="public"/>
</gateway>
</include>
Can someone over at skype support assist? or anyone else who has experience with this?
The 404 appears to be the more serious error but not sure if the 407 is causing the 404 and needs to be cleared up first.
Thanks! -
Hi I changed router 2 weeks ago still with PlusNet.My desktop Mac OS 10.8 defalts to BTWiFi hotspot login page every time I try to acess the internet.I have to go to the internet and manually choose PlusNet.I lock the page but the same thing happens the next time .There are 4 computers in the house and this is the only one doing this.What do I do
thank you
Geraldine HWHi,
interesting story. Did you find a way to configure phone for Internet calls (VOIP)? Using Setup/Connection/Admin Setup/SIP Setup (I am not sure about names as I am using Polish version) I may configure SIP profile and even register it ... but that is all. No way to make Internet calls even manual is saying about way of making them
regards, -
I have these items (9k, 400, pap2t) working together, however, when calling a spa voip phone (spa942) from a phone attached to the pap2t, I am unable to end up in voicemail if the user at the spa942 does not answer. From other SPA phones it is not a problem. I know there are issues with generic SIP clients getting into voicemail, but does that include the scenario I describe above? Do any of the spaxxx ATA's other than the pap2 have the ability to get forwarded into VM if an extension does not answer? Thanks.
I wanted to post a reply from another forum to what I had asked here. Looks like Cisco says ATA's won't be able to leave VM. You had indicated it should be possible....have you seen it working? Does the explanation below make any sense? Is there a different configuration that might work?
1. Feb 27, 2009 4:50 PM in response to:
Re: spa9k spa400 and pap2t
Hi,
I simulated your environment in my lab and ran a sniffer to capture and analyze the network traffic prior to answering here.
The issue is that the SPA9000, SPA400, SPA9xx phones, and WIP310 phones are a tightly coupled solution.
Non-SPA/WIP310 phones and ATAs do not supply the correct information in their SIP INVITE to the SPA9000 when attempting to leave vmail for the SPA942 on the SPA400.
SPA/WIP310 phones include a P-Mailbox: <mbox-ID>
in the SIP INVITE to the SPA9000 when attempting to leave vmail. [sip:vm@<SPA9000_IP>:6060]
In contrast, the PAP2T's analog phone sends a SIP INVITE to the SPA9000 but does not include the P-Mailbox: <mbox-ID> information because the PAP2T, nor any other ATAs are designed to work in the SPA9000 Voice System solution.
In summary, because the PAP2T's SIP INVITE to the SPA9000 to sip:vm@<SPA9000_IP>:6060lacks P-Mailbox: <mbox-ID> information, the SPA9000 responds with a 503 Service Unavailable and the analog phone caller finally hears a busy tone when the call is dropped.
Regards,
Patrick
Maybe you are looking for
-
Using Local File instead of SQL Databas
Some time ago I developed an application that could perform certain operations with the help of an SQL Database. It was just some few simple functions such as adding/removing and fetching data from the server. The thing is that the one that I was del
-
Budget checking for Q Category
Hi Experts, We are using WBS Elements in Procurement Process, When we creating PR with account assignment to WBS with Category P System is checking Budget and availability control is working fine but when i Creating PR against the AA Category Q Syste
-
See my comments made on iTunes when reviewing the app... The iOS version and the Mac version are now very out of sync. The new features all seem to be getting added to the iOS version including RAW support and video support. The Mac version does not
-
HAL seems to have broken after update. (SOLVED)
I'm not sure what happened, but after updating arch recently CD's, DVD's, and USB drives/sticks are not available. They "seem" to mount themselves as before, as they show up in gnome, but they are inacessible, clicking on any icon for them brings not
-
Why are PSE9 adjustment layers an improvement?
PSE6 has access to blend and opacity controls in adjustment layers but that's not so in PSE9, unless I'm somehow missing the controls. If that's the case, those controls are absent, how are the PSE9 adjustment layers an improvement?