Audio drop on Spectralink Wireless IP Phone with AP1200/350

I am currently using spectralink WLAN phone with Cisco AP1200/350,
I found audio would be dropped. It is caused by spectralink phone would
re-authenticate/re-associate AP many times in a short period time, Every time
it would casued audio dropped.
Any one experienced this kind of problem?

On a recent installation, we had terrible voice quality problems with CallManager-controlled Spectralink phones associated with AP1200s. The wireless LAN was in great shape for all other devices, and everything else like the SVP server was in good order.
We then set the "Data Beacon Rate" (DTIM) to 2, in accordance with Spectralink's recommendations. Everything started working well. I cannot tell you much about what that parameter is or does, but it worked for us. If you search Spectralink's website, there are PDF documents with recommended settings for various Cisco access points.

Similar Messages

Audio drops when using Three-Way Color Corrector in CS6

Hello all,
I recently encountered a very odd issue while editing a friends wedding video using Adobe Premiere Pro CS6. I had an interview with a guest laid over a music track and had all the audio adjustments made to both audio tracks so that the levels were where I wanted them. I then added the Three-Way Color Corrector to the video track and corrected the footage to my liking. When I played back this section of the wedding video, half way through the person's interview, his audio cut out. The audio from the music track was still playing, so it was not that audio drop that most people are familiar with within Premiere. I even muted the music track and played back the mentioned section of video and at the same exact point in time his audio just cuts out completely. I checked to see if any weird key frames had been added and nothing had been touched (that I could see). Also, the audio waveforms were still visible. I deleted the Three-Way Color Corrector effect off of the clip and suddenly there were no more audio problems. Eventually, needing color correction, I added an adjustment layer and assigned the Three-Way Color Corrector to that and had no issues, but I still don't see what could have been the problem, and it seems like a glitch to me. Has anyone ever had this happen to them? If so, what are your thoughts? Thanks!

The help does not tell you where the Tonal Range output now is as it has moved from the Output Menu.

Intermittent Audio Drop-offs with Airtunes to Apple TV

A number of other threads have similar problems. iMac 2.8 GHz iTunes 8.1 is connect to an updated ATV via new dual-channel ABS on the "n" band. Still there are audio drop offs with AirTunes as it did with the previous ABS Extreme n and previous version of iTunes. So much so, I just gave up and used the Apple Remote on the iPhone to play ATV. Any suggestions would be appreciated.

Switching from WEP to WPA/WPA2 personal seems to have totally cured the dropouts for me.
Also I used an old flying saucer Airport as the receiver via ethernet out for my entreatment center instead of using the AppleTV's wireless.
Also I had a Panasonic security cam that didn't support WPA so I used another flying saucer Airport via ethernet out as its receiver.
So now my setup is,
Airport Extreme in my home office as my main base station with a Motorola ADSL 2200 router feeding it internet.
Flying saucer Airport via ethernet and a hub feeding my AppleTV, DirecTV DVR, and a Mac mini.
Another flying saucer Airport via ethernet feeding my Panasonic security cam.
Airport Express for my bedroom speakers.
All airports are in bridge mode with the Motorola 2200 DSL modem acting as my router. (this seems to be the only way my Vonage adapter recovers without having to be reset if my DSL goes down)
I steam music from my 17" Macbook Pro to the AppleTV and Bedroom speakers all day now without any dropouts.

Audio drops problem iTunes with AE

I set up my AE to my hifi system. When playing music from iTunes, audio drops after 2-3 seconds. When I play a movie in iTunes, sound is fine.
I guess the problem has probably got something to do with sample frequency or bit rates. Any ideas?

I had similar problems with AE. Here my solution:
I have a French (Livebox) DSL modem with Wifi support. My MiniMac is connected to the DSL modem with an ethernet cable and Airport (did not check if I really need both). I have a MacBook connected over Wifi. The MacBook runs Leopard and iTunes 8. I configured the AE through the MacBook. This configuration worked only by using the ethernet cable connection between AE and MacBook. Once configured using the Airport utility with "Wireless Mode'='Join a wireless network' and using a 'manual' configuration, the AE booted correctly up to the static green LED. Streaming audio with iTunes worked corretly in this configuration.
I then wanted to stream audio from my MiniMac running Tiger and iTunes 8. I had the same problems as reported. 2-3 minutes correct working, then yellow blinking LED and again 2-3 green static LED and audio, then again yellow blinking LED.
Goggling sometime I found a site from which I could download older iTunes versions. I deinstalled iTunes8 on my MiniMac and installed iTunes7.4. That solved my problems! I can now stream audio from my MiniMac to AE without interruptions.Conclusion: Leopard + iTunes8 works, Tiger + iTunes8 does not work, Tiger + iTunes7.4 works.

I have a wireless home phone connect device. Can call out on it when dial tone present, but most times cannot receive calls since purchase. Anyone else having a problem with their home connect device?

Have had Wireless Home Phone connect device refreshed and updated three times since I purchased it in September 2014. Last tech I worked with on it initialized an NRB to check the network. I checked the location of the tower, which is less than one mile up the hill with clear line of site. The device always has two antennas on it. The device is great when it does work, but most of the time it rings three times and then jumps to an automated message saying the person I am calling is not available. I've actually gone up there several times to try an fix the device, but always end up calling a Verizon tech who has me unplug it, pull the battery, and they do some network stuff and then we turn it back on and it works. After the last time on 10/02/14, I called it and got the message. I drove up there, 100 miles one way, and called it while standing in the room. Got the same message. I picked up the phone, heard a dial tone and dialed my phone and it range through. I hung that phone connected to the device and call it using my cell phone. It rang through. I went home and tried to call it the next day. Again it did not ring through and I got the message after three rings. I call another phone at the facility and had them call me on my mom's phone and it is fixed it again. The fix last for a day or two and then it's broken again, like it is today. If it's a network problem then it must be fairly close to the area because no matter what we do it fails again. So much for the 99.98% reliable. This thing is a piece of junk!
I purchased this device and put it on my plan so that my mother who we placed in a home at 96 years old would have a phone connection that she could use and wouldn't flip her out. My family and her family has not been able to faithfully connect with her at this critical time in her life. I am giving Verizon 10 more days to figure this out and then I am going to ask for a total refund not to mention how much time and travel I have put into this piece of junk. Either the home connect transceiver is broken or the network is broken. Why should I pay close to $70 for a service that has not worked more than several days out of 35 days? Also I have spent close to three hours on the phone with decent technical people, but it's my time. I should send Verizon a bill for every time I have gone up there to fix this device as well.
What perplexes me is that a regular cell phone would have sufficed, but because Verizon has to have a data connection that is not available on a regular phone and it was going to cost me $30/month, I decided to go with the home connect. Seemed like a nice solution for an older person who would have a hard time using a cell phone and keeping it charged. Plus after all the fees for getting it connected, it only cost $20/month. That would be a pretty good thing, IF IT WORKED....... The other thing that irks me is, why do I have to pay for a data connection when the cell phone Verizon originally provided can't even access the network. What ever happened to adding a phone line for $10 bucks\ month? All I wanted was a simple phone. She would not be able to use anything more complex and all she needs is a phone line period.
I have no problem paying for something that works and does what it should, but this is ridiculousness. The MTF on this device is somewhere around 1 to 2 days. Out of 365 days that's about a 96.5% failure rate....... Want my advice? Get ride of this device. It sucks and it's bad marketing.

jsavage9621,
It pains me to hear about your experience with the Home Phone Connect. This device usually works seamlessly and is a great alternative to a landline phone. It sounds like we've done our fair share of work on your account here. I'm going to go ahead and send you a Private Message so that we can access your account and review any open tickets for you. I look forward to speaking with you.
TrevorC_VZW
Follow us on Twitter @VZWSupport

Audio drop-out in FCPX with 1920x1080 & 2048x1240p @ 24p (codec Apple ProRes 4444 (Linear PCM))24p

Hi all...
I'm new to this forum.
Is there an audio issue (drop-outs) with FCPX (ver 10.1.2) for 1920x1080 24p, 1080p HD & 2048x1240 24p (2k) - note: films in FCPX are in codec Apple ProRes 4444, Linear PCM?
Very recently I've been working on a show reel and I have imported a file (short film) to snip out a section to add to the show reel.
I set the project to the film's settings i.e.. "set based on first video clip"...(1920x1080 24p, 1080p HD). When I play the film in FCPX the audio "hic-cups"..."dips to silence"..."drops-out" briefly (up to 0.5 secs in duration) and very frequently all the way through (approx 15 mins long film)...note: audio set is at stereo 48kHz. The audio monitors on the right hand side also show the dip taking place (if it's a long enough drop-out).
I also have a version of the film in 2048x1240 24p at 2k, again Stereo 48kHz...and this does the same audio "drop-out" all the way through. I'm unsure what other formats will create this problem i.e. higher or lower aspect ratios than those above.
Is this a software and/or OS issue? Or a hardware issue? Or both!?
About This iMac details:
iMac 27 inch: OS X ver 10.9.4
Processor: 3.5 GHz Intel Core i7
Memory: 32 GB 1600 MHz DDR3
Graphics: NVIDIA GeForce GTX 780M 4096 MB
Fusion Drive 3.12 TB
Final Cut Pro X was bought & installed with this iMac back in Feb 2014
FCPX version: 10.1.2
Note also: I used Compressor (ver 4.1.2) to convert the files (movies) above to Apple ProRes 422 HQ, Linear PCM and the issue disappeared - the audio and visuals played with no problems. All films play with no issue when opened in QT player. They are also fine when exporting from FCPX. The issue only occurs (as far as I can tell) within FCPX when editing or playing the audio there.
Has anyone encountered this problem?
Kind Regards,
Hakan

Hi Luis,
Thank you for that.
Seems like you have a point about the "hardware" issue. I'm unsure (new to it all) at what data rate my (er...wonderful) fusion drive provides FCPX for smooth playback - or where I can find that info on the iMac. Playing back movies in the 422 HQ codec seems to have no issues (when I converted the movie from 4444 HQ).
But playing the full 4444 HQ, the picture is smooth, but the audio has these drop-outs - even with "best performance" option ticked. Interesting that the audio suffers, not the picture in FCPX - so not sure what's going on there. I also read - perhaps - that to play a smooth ProRes 4444 HQ movie in FCPX the data rate should be min of 500 Mbs (based upon About Apple ProRes) for HD movies and up (i.e. 2k). ProRes 422 HQ needs a data rate of 220 Mbs - so that is my upper-limit to smooth video&audio playback with the current system I have, I guess.
Does anyone recommend a fast USB3 or Thunderbolt drive (500Mbs+) where I am able to run 4444HQ movies with no issues to picture or sound (away from the fusion drive)?
KR
H.

Since i got the new Iphone 4s it constantly drops calls. Apple have replaced the phone with another new iphone 4s but i am still having the same problem. I have checked with vodapohe that there is no issue from there end, given a new sim, still drops out

Since i got the new Iphone 4s it constantly drops calls. Apple have replaced the phone with another new iphone 4s but i am still having the same problem. I have checked with vodapohe that there is no issue from there end, given a new sim, still drops out

Have a look at this it might help
http://support.apple.com/kb/TS4148

Is there a way to pair more then one set wireless head phones to either a iphone or ipad with the blue tooth?

Is there a way to pair more then one set wireless head phones to either a iphone or ipad with the blue tooth?

Yes. I have two different Bluetooth headsets and they are both paired to my iPhone so I can use either one but only one can be used at any one time.
To pair the second one, simply repeat the pairing process that you used for the first one.

HT1386 When I sync my i-phone with my pc, will it pull audio from my phone to my i-tunes on my pc? My problem is I have songs on my pc that will not play - the message says "original file could not be found"... They play on my phone.

I am not sure if the synching will pull the audio from my phone and replace the 'not working' files on my i-tunes, or if it will replace the working audio on my phone with the non-functioning audio on my pc.
Thanks for any insight...

No it won't. You are getting that message because those song are no longer in your my music folder and itunes doesn't know where to look for the file.

ITunes scratchy he||...with audio drop outs

I use iTunes v7.4.2 all the time to play music. Only lately my iTunes has been sounding very intermittently scratchy (static-y) and the audio drops out (without the diamond moving) at unexpected intervals (anywhere from 10 sec to several minutes). When I try to hit the volume buttons, I don't get any feedback sounds like usual. Waiting it out seems to be the only remedy...then it comes back again, and again...
I use MenuMeters to monitor my CPU and bandwidth speed and so I notice that this happens most when I am downloading torrents. Very frustrating! Curiously, I tried running a bootable clone of my system on a separate drive and I found that the problem went away (I'd have to run it for longer to be sure though). Could this be a sign of imminent hard drive failure?
Has anyone else experienced these issues? Does it sound like a hardware or software issue?
Thanks for any replies,
-Bill
Message was edited by: BillytheKidney

I am going to keep researching this issue for you.
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Audio drops with the 2 gig file limit size between clips

Videoing with a Sony HXR-NX5U, the camera creates a 2 gig file size limit then importing the clips into Premiere Pro CS6. There is a few frames between clips where the audio drops and there is also s very slight drop in the video, this happens only between longer clips. I saw some reference to this in the forum a while back but no answer. Any indeas!!

Simple fix - copy entire contents of SD card to new folder on hard drive, all folder intact. Then in Premiere, import using "Media Browser" rather than "File > Import". With Media Browser, the spanned clips will be properly handled, importing as a single, long clip with no glitches.
Thanks
Jeff Pulera
Safe Harbor Computers

CUCM 8.6 Dropped call transfers involving SIP phones

Hi All,
I am a developer who has been tasked with figuring out why call transfers are being dropped by Cisco CUCM when the original call comes from a SIP phone. This scenario works:
Cisco phone calls another Cisco phone, which transfers the original call to a SIP phone
These scenarios do not work:
SIP phone calls Cisco phone, which transfers the original call to another Cisco phone
SIP phone calls Cisco phone, which transfers the original call to another SIP phone
I have researched the Call Manager traces to the best of my ability, and I see some info in there that could potentially point to the source of the problem. I am just unable to understand what the trace means:
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/active_CcDisconnReq: ccDisconnReq.onBehalfOf=Media : ccDisconnReq.s.sv=2 : ccDisconnReq.c.cv=47 |1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/Stack/Info/0x0/sipConstructContainerContext #### Created container=0xb0b42f58|1,100,71,1.1^*^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|1,100,63,1.93259^10.10.10.85^*
I have been wondering whether this could be a codec issue, however the SIP phones we are using are configured with the following codecs:
G711U
G711A
G722
ILBC
GSM
and our SIP software is also set to accept the first codec offered by the remote side. It seems from the SIP client logs that G722 is being used as the codec to communicate with the Cisco phones, but perhaps I'm misinterpreting.
I have attached a CUCM trace of a call from a SIP phone (ext. 491) to a Cisco handset (ext. 170) where the Cisco handset attempts to transfer the call to another SIP phone (ext. 492). The trace snippet shown above is from this log.
I would really appreciate it if someone more experienced with VoIP/SIP/CUCM could take a look and offer any ideas on what the issue might be, and also how we might be able to address it. I can try to provide more info about our CUCM configuration if needed.
Thanks in advance!

Leslie, so here is what I found from the traces....
To understand the difference we need to understand how cucm performs call transfers from a sccp signalling point and a sip signalling point
SCCP
When the transfer key is pressed
1. CUCM sends a CloseReceiveChannel and StopMediaTransmission to the IP phone involved in active media (referenced by the callids)
NB, here CUCM updates the call state on the phone to a call state of 8 which is "Hold"
2.CUCM tells the held party to listen MOH from MOH server
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5..CUCM sends a CloseReceiveChannel between the held phone and MOH server (to tear down the media)
6. Next CUCM sends a CloseReceiveChannel and StopMediaTransmission to the transfering party & transfered party to remove Xferring party from call
7. finally CUCM sends OpenReceiveChannel between the original called party and the transfered party..and call is done
For SIP signalling. when the first transfer key is pressed
1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
2. CUCM sends a Delayed offer to insert MOH or to resume Media stream
NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
6.Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call
Now having explained all of these, we need to look at where the call is failing for SIP-----SCCP----SIP calls without MTP
lets look at succesful SCCP-----SCCP-----SIP without MTP
Point 4 above
++++++++Extension 170 presses the transfer button to connect the two calls (Callid=24378483)+++++++++++++
(0003395) SoftKeyEvent softKeyEvent=4(Trnsfer) lineInstance=1 callReference=24378483
Point 5 above
++++Next CUCM closed the media between extension 160 and MOH server callid=24378480(this is the only active call on this callid)+++
(0003396) CloseReceiveChannel conferenceID=24378480 passThruPartyID=16777845. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
Point 6 Above
+++++Next cucm closes the call between extension 170 and 490 callid=(24378483)++++++++
(0003395) CloseReceiveChannel conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
(0003395) StopMediaTransmission conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
Point 6 above for the sip side (since the destination is SIP, to tear down media to SCCP phone, so as to connect the caller to the xfered party)
+++++++Next CUCM sends a re-invite with a=inactive SDP to the sip phone ++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885626,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23332dbee978
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
o=CiscoSystemsCCM-SIP 192115 2 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0
m=audio 24560 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=inactive-----------------------------------------------------Inactive
Still part of Point 6 for SIP signalling
++++++++++++Next sip phone responds with a 200 OK recevonly SDP +++++++++++++++++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885628,NET]
SIP/2.0 200 OK
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
a=ptime:20
a=recvonly-------------------------------------a=recvonly
Finally Point 7 above..
+++++++++++++=Next cucm sends a DO re-invite to extension 492-sip phone++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885630,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
+++++++Next we get a 200 OK from sip phone with sdp=sendrecv+++++++++=
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885634,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
Contact:
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
Call-ID: [email protected]
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
m=audio 16574 RTP/AVP 9 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
+Now CUCM sends an OpenReceiveChannel and start media xmission to sccp phone (callid=24378480) with media parameters of sip phone++++++
(0003396) OpenReceiveChannel conferenceID=24378480 passThruPartyID=16777848 millisecondPacketSize=20 compressionType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierIn=? sourceIpAddr=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104). myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
(0003396) startMediaTransmission conferenceID=24378480 passThruPartyID=16777848 remoteIpAddress=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104)
remotePortNumber=16574 milliSecondPacketSize=20 compressType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierOut=?. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
+++++++++++=Next Phone sends its ACK+++++++++++++++
(0003396) OpenReceiveChannelAck Status=0, IpAddr=IpAddr.type:0 ipAddr:0x0a0a0a89000000000000000000000000(10.10.10.137), Port=20352, PartyID=16777848
+++++++++++=Next CUCM sends ACK to 200 OK from SIP Phone+++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885635,NET]
ACK sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23366067b8c0
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
Date: Tue, 19 Feb 2013 21:44:45 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.137
b=TIAS:64000
b=AS:64
t=0 0
m=audio 20352 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Now at this point all is well...and the call is connected....
Now here is where the call is failing on the SIP-SCCP-SIP call without MTP
From Point 2 above, CUCM sends a DO to insert MOH, and then gets response, then sends an ACK to 200 Ok back to SIP Phone..
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881160,NET]
ACK sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22035ecc1fcb
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:38:50 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Max-Forwards: 70
CSeq: 102 ACK
o=CiscoSystemsCCM-SIP 190666 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------IP address of MOH server
t=0 0
m=audio 4000 RTP/AVP 0--------------------------------MOH port 4000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendonly---------------------------------------------------------sendonly
+++++NOW Point 6 above (SIP) CUCM sends a=inactive to break media path to MOH server to connect caller and xfered party++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881161,NET]
INVITE sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:39:04 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 164
v=0
o=CiscoSystemsCCM-SIP 190666 4 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
t=0 0
m=audio 4000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=inactive---------------------------------------------------------------------media inactive
At this point, we should get a response back from the sip phone...
and here is what we got..
++Trying which is expected++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 331 bytes:
[881162,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 103 INVITE
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Content-Length: 0
++++++++Then we get a BYE+++++++++++++++
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 576 bytes:
[881163,NET]
BYE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.104:53361;branch=z9hG4bKa2vdQvR7J9OiMyjU;rport
Contact:
Max-Forwards: 70
From: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.8
To: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 3 BYE
Content-Length: 0
So this is the root cause of the problem. Your SIP phone does not know how to respond to multiple media break between it and the MOH server.
The difference between this and the succesful SCCP-SIP--SCCP, is that the held party was a sccp phone, hence the sip phone only has to process one a=inactive SDP message, where as in the SIP-SCCP-SIP, the help party was sip, so the sip phone has to process two a=inactive SDP messages
Now what is the difference when MTP is involved! A Big difference. MTP stays in the media path. So there is never a break in media and no inactive SDP attribute is sent. The flow looks like below:
for the initial call...The SIP phone sends its media to MTP and likewise the SCCP phone
SIP------Media------MTP------------Media-------SCCP Phone
When the new destination is dialled and transfer is commited,
SIP-------------media----MTP--------media---------MTP
The final invoite sent to connect 492 and 491 has MTP as the IP address to connect Media to.
++++++++Ivite to 492 ++++++++++++++
INVITE sip:[email protected]:61303;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231a3b24b862
From: ;tag=192048~d8e94532-127d-4dca-bba0-64b1675da032-24378472
To: ;tag=78FF5BF6C019A55EA020B69BB6A767E2
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 214
v=0
o=CiscoSystemsCCM-SIP 192048 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------------------------------the MTP ip address
t=0 0
m=audio 25038 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++Invite to 491 +++++++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.94 on port 50376 index 1887
[885429,NET]
INVITE sip:[email protected]:50376;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231b78d1b56
From: ;tag=192046~d8e94532-127d-4dca-bba0-64b1675da032-24378467
To: "491" ;tag=F13CE94DE942C47680356A647DC7F916
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: AE7045FFB2D8D9C28D54651473A14A5D41B5B93C
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: "Leslie2" ;party=calling;screen=no;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192046 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195----------------------------------------MTP
b=TIAS:64000
b=AS:64
t=0 0
m=audio 25030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Wao! That was a long one isnt it...It was fun too.
So now you can look at your sip phones and see if they can accept two inactive sdp messages within the same call. That way you can remove MTP. otherwise you will have MTP involved in every single call involving a sip phone, even if they do not involve transfers
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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Unified Mobility, no audio on Send call to Mobile Phone

I'm using UCM 9.1 with Unified Mobility (xfer to alternate number) with good success if I follow a typical call flow:
Inbound call -> Ext -> Rings deskphone x seconds -> Rings mobile phone -> Answer mobile phone -> hangup -> Resume call on desk phone.
But if I pick up a call on my desk phone, and use the Mobility button to xfer a call to my mobile phone I get no audio:
Inbound call -> Ext -> Pickup desk phone -> Mobility soft button, 'Send call to Mobile Phone' -> Answer mobile phone, no audio -> Hang up mobile phone -> Resume call on desk phone (two-way audio).
Device wise the call flow is:
ITSP SIP trunk -> CUBE -> CUCM -> 7965 IP Phone.
Recently I reconfigured CUCM to use the CUBE for any MTP resources instead of the software option and I think I may have missed something.
CUBE config:
voice-card 0 dspfarm dsp services dspfarm!!!voice service voip ip address trusted list ipv4 173.46.30.218 ipv4 173.46.30.202 ipv4 10.0.6.30 ipv4 10.0.6.31 ipv4 10.0.6.33 ipv4 10.0.6.32 ipv4 10.1.1.4 ipv4 10.0.250.0 255.255.255.0 mode border-element allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip redirect ip2ip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip midcall-signaling passthru media-change early-offer forced no call service stop registration passthrough!voice class codec 1 codec preference 1 g711ulawsccp local GigabitEthernet0/0.42sccp ccm 10.0.6.30 identifier 1 version 7.0 sccp!sccp ccm group 1 bind interface GigabitEthernet0/0.42 associate ccm 1 priority 1 associate profile 1 register MTP_2951-01!dspfarm profile 2 transcode universal codec pass-through codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 11 associate application SCCP shutdown!dspfarm profile 3 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 associate application SCCP!dspfarm profile 1 mtp codec pass-through codec g711ulaw maximum sessions hardware 15 associate application SCCP!
In CUCM I removed the software MTP from the MRG:
Not sure where to start troubleshooting this problem, any help is appreciated.
Steve

Thanks for the help gents. I couldn't get to this until we're out of office hours on the weekend.
Interestingly, I have no mid-call option in my dial-peers. This is a 2951 running 15.2(4)M2.
I double checked the MRGL, my phone is associated with it.
Codec is G711 on ITSP side, and on phones - I'm not sure I fully understand the use cases for MTP, this is something I need to research more.
I've included two ccsip message debugs, the first one is the existing issue of no audio (in either direction).
The second I've changed the midcall-signaling passthru option, dropping the media-change bit and we get audio in both directions for mobility except we use Unity call handlers for IVR functionality, and now when an inbound caller is forwarded to an extension we get no audio - obviously this is a game stopper.
In Unity I have the port group configured as SCCP - Maybe I should be using SIP instead?
No Audio:
voice service voip
sip
midcall-signaling passthru media-change
early-offer forced
no call service stop
registration passthrough
Dec 8 21:07:35.842: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 101 INVITEExpires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationCisco-Guid: 3244526336-0000065536-0000003600-0503709706Session-Expires: 1800Diversion: ;reason=follow-me;privacy=off;screen=yesP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: ;isFocusMax-Forwards: 70Content-Length: 0Dec 8 21:07:35.850: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 3244526336-0000065536-0000003600-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITETimestamp: 1386536855Contact: Expires: 180Allow-Events: telephone-eventMax-Forwards: 69Diversion: ;privacy=off;reason=follow-me;screen=yesContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 7885 9952 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29558 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:35.850: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:35.858: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Dec 8 21:07:41.986: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 253v=0o=root 959120698 959120698 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 40818 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:07:41.986: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Length: 0Dec 8 21:07:41.990: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 183 Session ProgressVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3614 6206 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29556 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:41.990: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 180 RingingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Server: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:43.938: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 253v=0o=root 959120698 959120699 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 40818 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:07:43.938: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Require: timerSession-Expires: 1800;refresher=uacSupported: timerContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3614 6206 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29556 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:43.942: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274C4F2From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: telephone-eventContent-Length: 0Dec 8 21:07:43.958: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1d288c5e53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 218v=0o=CiscoSystemsCCM-SIP 40265 1 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.6.30t=0 0m=audio 4000 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=inactivea=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:07:44.158: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: UPDATE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1f50a2749fFrom: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: Cisco-CUCM9.1Max-Forwards: 70Supported: timer,resource-priority,replacesAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 102 UPDATESupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:07:44.158: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1f50a2749fFrom: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 102 UPDATEAllow-Events: telephone-eventContact: Supported: timerContent-Length: 0Dec 8 21:07:44.162: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2028eab237From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 103 INVITEMax-Forwards: 70Expires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationSession-Expires: 1800;refresher=uacP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:07:44.162: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2028eab237From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: 103 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:44.162: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2028eab237From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: 103 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Require: timerSession-Expires: 1800;refresher=uacSupported: timerContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3614 6206 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29556 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:44.214: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2151dd5c40From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: 70CSeq: 103 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 232v=0o=CiscoSystemsCCM-SIP 40265 3 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.250.93b=TIAS:64000b=AS:64t=0 0m=audio 25834 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:07:52.590: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK76uf9710bomh2kk6c350.1Max-Forwards: 68From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Call-ID: [email protected]: CSeq: 102 INVITEUser-Agent: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 253v=0o=root 959120698 959120700 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 40818 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:07:52.594: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK76uf9710bomh2kk6c350.1From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: 102 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:52.594: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK76uf9710bomh2kk6c350.1From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: 102 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 7885 9953 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29558 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:52.610: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK8m4hbg10c8ag4kg723g0.1Max-Forwards: 68From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Call-ID: [email protected]: CSeq: 102 ACKUser-Agent: Rogers SIP CoreContent-Length: 0Dec 8 21:07:52.630: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: BYE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK96bidp10cou05l89e611.1Max-Forwards: 68From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Call-ID: [email protected]: 103 BYEUser-Agent: Rogers SIP CoreX-RBS-SIP-HangupCause: Normal ClearingX-RBS-SIP-HangupCauseCode: 16Content-Length: 0Dec 8 21:07:52.634: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK274D1192From: ;tag=475844D8-1CC2To: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2Max-Forwards: 70Timestamp: 1386536872CSeq: 101 BYEReason: Q.850;cause=16P-RTP-Stat: PS=0,OS=0,PR=420,OR=67200,PL=0,JI=0,LA=0,DU=8Content-Length: 0Dec 8 21:07:52.634: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK96bidp10cou05l89e611.1From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 103 BYEReason: Q.850;cause=16P-RTP-Stat: PS=2,OS=320,PR=100,OR=16000,PL=0,JI=0,LA=0,DU=8Content-Length: 0Dec 8 21:07:52.646: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK274D1192From: ;tag=475844D8-1CC2To: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: 101 BYEContent-Length: 0
bi-direcitonal audio:
voice service voip
sip
early-offer forced
midcall-signaling passthru
no call service stop
registration passthrough
Dec 8 21:09:44.331: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 101 INVITEExpires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationCisco-Guid: 0239559040-0000065536-0000003601-0503709706Session-Expires: 1800Diversion: ;reason=follow-me;privacy=off;screen=yesP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: ;isFocusMax-Forwards: 70Content-Length: 0Dec 8 21:09:44.339: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0239559040-0000065536-0000003601-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITETimestamp: 1386536984Contact: Expires: 180Allow-Events: telephone-eventMax-Forwards: 69Diversion: ;privacy=off;reason=follow-me;screen=yesSession-Expires: 1800Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 4519 2507 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29562 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:44.339: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:09:44.347: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: Call-ID: [email protected]: 101 INVITETimestamp: 1386536984Dec 8 21:09:52.535: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 101 INVITETimestamp: 1386536984Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 255v=0o=root 1961674502 1961674502 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:09:52.539: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 183 Session ProgressVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:53.007: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 101 INVITETimestamp: 1386536984Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Length: 0Dec 8 21:09:53.007: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 180 RingingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Server: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:09:54.967: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 101 INVITETimestamp: 1386536984Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 255v=0o=root 1961674502 1961674503 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:09:54.967: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:54.967: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274F1060From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beDate: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: telephone-eventContent-Length: 0Dec 8 21:09:54.979: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2876c28ab9From: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 218v=0o=CiscoSystemsCCM-SIP 40276 1 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.6.30t=0 0m=audio 4000 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=inactivea=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:09:55.007: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: UPDATE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2ada930ebFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: Cisco-CUCM9.1Max-Forwards: 70Supported: timer,resource-priority,replacesAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 102 UPDATESupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:09:55.011: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2ada930ebFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 102 UPDATEAllow-Events: telephone-eventContact: Supported: timerContent-Length: 0Dec 8 21:09:55.011: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2be82180dFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 103 INVITEMax-Forwards: 70Expires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:09:55.011: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK275014C3From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beDate: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0239559040-0000065536-0000003601-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 102 INVITEMax-Forwards: 70Timestamp: 1386536995Contact: Diversion: ;privacy=off;reason=follow-me;screen=yesExpires: 180Allow-Events: telephone-eventContent-Length: 0Dec 8 21:09:55.011: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2be82180dFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 103 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:09:55.019: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK275014C3From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 102 INVITETimestamp: 1386536995Dec 8 21:09:55.031: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK275014C3From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 102 INVITETimestamp: 1386536995Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 279v=0o=root 1961674502 1961674504 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:09:55.035: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2be82180dFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 103 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:55.175: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2c8e71c96From: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 70CSeq: 103 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 232v=0o=CiscoSystemsCCM-SIP 40276 3 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.250.93b=TIAS:64000b=AS:64t=0 0m=audio 22546 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:09:55.179: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK2751A88From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beDate: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 70CSeq: 102 ACKAllow-Events: telephone-eventContent-Type: application/sdpContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 4519 2508 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29562 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:10:05.267: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK3tuoo01070ag0lg7o5k0.1Max-Forwards: 68From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Call-ID: [email protected]: CSeq: 102 INVITEUser-Agent: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 255v=0o=root 1961674502 1961674505 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:10:05.271: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27523EAFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0239559040-0000065536-0000003601-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITEMax-Forwards: 70Timestamp: 1386537005Contact: Expires: 180Allow-Events: telephone-eventContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:10:05.271: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK3tuoo01070ag0lg7o5k0.1From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 102 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:10:05.275: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27523EAFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: presenceContent-Length: 0Dec 8 21:10:05.275: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27523EAFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYAllow-Events: presenceSupported: replacesSupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=called;screen=yes;privacy=offContact: Content-Type: application/sdpContent-Length: 232v=0o=CiscoSystemsCCM-SIP 40276 3 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.250.93b=TIAS:64000b=AS:64t=0 0m=audio 22546 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:10:05.279: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK3tuoo01070ag0lg7o5k0.1From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 102 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 4519 2508 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29562 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:10:05.279: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27531E8DFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: telephone-eventContent-Length: 0Dec 8 21:10:05.295: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK4d5qq910785h6ks9f3c0.1Max-Forwards: 68From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Call-ID: [email protected]: CSeq: 102 ACKUser-Agent: Rogers SIP CoreContent-Length: 0Dec 8 21:10:05.295: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: BYE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK4tbtsi10785h6jcp71g1.1Max-Forwards: 68From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Call-ID: [email protected]: 103 BYEUser-Agent: Rogers SIP CoreX-RBS-SIP-HangupCause: Normal ClearingX-RBS-SIP-HangupCauseCode: 16Content-Length: 0Dec 8 21:10:05.295: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK275469CFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2Max-Forwards: 70Timestamp: 1386537005CSeq: 102 BYEReason: Q.850;cause=16P-RTP-Stat: PS=511,OS=81760,PR=505,OR=80800,PL=0,JI=0,LA=0,DU=10Content-Length: 0Dec 8 21:10:05.299: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK4tbtsi10785h6jcp71g1.1From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 103 BYEReason: Q.850;cause=16P-RTP-Stat: PS=505,OS=80800,PR=634,OR=101440,PL=0,JI=0,LA=0,DU=10Content-Length: 0Dec 8 21:10:05.303: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK275469CFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 102 BYEContent-Length: 0

Airport express - audio dropping

i'm using airport express for wireless internet and to play audio through a pair of powered speakers. it has worked fine in the past but lately has been dropping the audio constantly. i'm playing music through itunes and the track just stops, and the audio ceases. i'm only about 1 meter away from the airport.
any ideas? thanks!

Same problem here. The audio drops seem to directly correlate with any and all web activity in Safari. Every time a new page has to load, Airport Express cuts out. I've had Airport Express (n) for a few months and only recently did this start happening.

CPU peaks / audio drops (here comes another one)

Hi all,
I know this problem exists (sudden CPU-peaks and audio-drops as a result), but I wonder if anyone has experience with M-Tron Pro in Mainstage and these audio-drops.
I'm having those problems and they're really a pain in the ***. Can't use it live because of the drops and I really need the M-Tron sounds.
Airport or wireless networks are all turned off, Mainstage (latest version 2.1.2) is the only programming that's running, CPU-meter shows a default usage of about 10% and yet I keep getting regular audio drops/CPU-overloads when I use the M-Tron Pro plugin.
Strange thing is: in Logic I can keep playing the M-Tron for ages without one single audio drop... One wonders why the internal organization or CPU-management of two programs based on the same "logic" (or at least it seems) is so different?
No other solution for me than learn to work with Ableton and run a stable live system then? Or other suggestions here?

I ran into similar problems when using a 2.0G machine, now on a 2.4, and things improved greatly. A couple of things to try:
1) run your Activity Monitor utility to see if there is anything running in the background you aren't aware of
2) try downloading and running smcFanControl to run your fan at a higher rpm when using MainStage
Have you definitely isolated the culprit to M-Tron? (after eliminating other suspects, plug-ins, etc). If so, are there any ways to minimize the CPU hit within M-Tron (check manual and forums)? I don't have the software, so can't check. For example, I use Superior Drummer, and found a lot of CPU and memory reducing tips, which helped load greatly.

Audio drop on Spectralink Wireless IP Phone with AP1200/350

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