My IP4 drops calls alot call failure is what it says on screen even in good service area and good bars on phone

My iphone seems to drop alot of calls and has done for the past couple of months.
It has the latest software update
It does it even when in good mobile phone coverage
You can call straght back
Dont know if it is the phone or the reception
Is there anything i can do to check

It sounds like an O2 problem, you'd have to get in contact with them on 0800 977 7337
Have you tried the SIM card in another phone?

Similar Messages

HT201232 I have called the support number given, was on hold for more than an hour and received quite unpleasant phone bill for trying to sort out AppStore mistake! Is there any other options of getting "support" apart from calling these telephone numbers

My account was charged for buying app that I actually didn't buy. I was trying to sort out these mistake by calling the support number given (Russia). There was no other options like sending an email or calling some other number. I was on hold for one hour and a half! Is that normal for Apple support??? So apart from being charged for the app I haven't given my permission to buy, I also got a phone bill for calling the only support number available. Where can I submit my complain and what could be done about it?
Thank you,
Natalia

There is an email way to do this. See here: https://getsupport.apple.com/ContactInfo.action
The Russian Federation is one of the countries listed under Other.

Is there another option if you forget your answers to your security questions besides calling customer support. Please help I'm very awkward to explain things and especially over the phone.

what to do if you forget the answers to your security questions?

If you didn't set up a rescue email address you'll have to call them.

HT4623 in my my ipod when i go the general there is no icon called software update , what can i do ? even when i open it in itunes they says that software 4.2 is up to date i cannot download kik, instagram and the rest social network

i cannot update my software on my ipod

Identify iPod Models
http://support.apple.com/kb/ht1353
The 2nd Gen can only go to 4.2.1

Receiving"No Service" message and cannot make/receive phone calls

Receiving "No Service" message and cannot/make receive callsl

Have you contacted your cellular carrier?
Here's how to contact them.
Contact a mobile carrier

Agents in not ready state too long drops calls.

Using CRS 4.0(3) with agent desktop.
Issue is that when agents go to lunch they go into the not-ready state until they come back from lunch. Upon coming back from lunch, they proceed to go ready and then they start dropping calls as they come into the que.
If you log out completely and close the application when you leave for lunch versus going not ready for an extended period of time the issue does not appear when you log back in.
Any ideas?

Update on this. I looked at the control center and saw that my server was in partial service. I verified the service was the JTAPI subsystem.
I ran the JTAPI update client tool from this link
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_note09186a00801c8316.shtml
and saw the version of JTAPI between the CCM and the IPCC server were off. I updated both IPCC servers to the same version of call manager and will now hold to do some testing.

RAZR MAXX drops calls "on command" and randomly

I have a RAZR MAXX (two months old). If I use my Plantronics Pro HD Bluetooth device, the phone will suddenly drops a phone call without any reason. If I don't use the BT device and use only the phone/speaker, everything is working fine. The same goes if I connect another BT device as well as a wired earpiece. It suddenly gives the loud beep in the ear and the call gets disconnected. This happens 2-3 times per day out of 40 phone calls.
My boss has a RAZR MAXX as well, and he has the same identical problem... only today, he was able to "make" the phone drop the call by simply placing the phone back into the holster. He uses only a wired earpiece and the call would drop ONLY when the earpiece is connected. We tried a phone call by having the conversation on the speaker, and he was able to place the phone into the holster without any problem... the moment he connected the earpiece plug to the phone, I got the beep and the call got disconnected.
Can ANYONE confirm or had any experience with this crap?

You are absolutely correct. The person who deleted my posts AND his own post which made no sense whatsoever to the initial question I posted, is probably reading this post with interest, as I'm reporting this incident to Verizon's Admins.
Here is what my boss discovered in the last few days since I posted the question:
1. Him and I use a leather holster with a latching magnets to close the flap. When he placed his phone in the holster while on call, and closes the flap, the call gets disconnected... but ONLY if his $50 wired (forget Bluetooth for a moment) is connected to the phone. He now uses a piece of Velcro (ugly but works) to hold the phone in the holster. When the flap is open, he can use the phone WITH the wired earpiece without a problem. There is no question it is a design issue of the phone.
2. In my case, the fist call of the day today, which I used the speaker option, not even my BT, the call was disconnected after 3 seconds... It is definitely the phone. This is without turning the BT ON...
So far, the only way MY phone is working properly as a phone is when I bring it close to my head and talk... For free cancer catalyst I can go out to the desert and enjoy the sun... The phone is a pain AS A PHONE.
3. I know for sure it is NOT Verizon. I'm with the company for year and used BlackBerries until the RAZR MAXX... Maybe I'll need a PHONE phone AND and RAZR MAXX as a data device... so why did I buy the tablet?....
So far, only by playing with the phone things work... But I don't have time to play with the phone every day just to make some calls and mainly to receive many calls....

Suddenly dropping calls

How do I report trouble with dropped calls, spotty service?
Used to get perfect service here at home and suddenly in the last 1-2 weeks I cannot talk to anyone when I am at home, it braeks up and drops calls.
I have the old iphone 4. Had great service from November 2012 until just now. No case on the phone, has not been dropped or anything like that.
Is this a tower issue? If so, how do I contact someone about that? Or is it an issue with my phone.

Same thing happening with me over the last few weeks. I also cannot send texts with pictures. The same thing is also happening with one of my neighbors who has an iphone5. Is there any help with this. I tried *228 and that did not work. Is this a network overload issue?

7.0.4 dropping calls

Anyone else experiencing dropped phone calls (not facetime) since this most recent update?

I purchased an iphone5 a week ago and it dropped calls steady(case or not)I could not even recieve calls.Took it back to place of purchase,they said its your sim card,I said dont think so,I was right,two completed calls.then same old same old.Took me 4hrs of online research till I found my cure(3 days and not one droppred call) so far. First I did a factory reset(after deleting all music,photos and downloaded content such as apps,ringtones,wallpapers etc) then a restart.Secondly I turned my time set/date/location to manual and set all three(found at settings-general-date and time) restart.Third I turned off caller id/call forward/call waitng(found at settings-phone) restart.Fourth I turned off passcode and screen lock(found at settings-general) restart.Fifth I turned all my push notifications off(found at setting-notification center) restart and so far so good.I did turn my caller id back on and so far so good.It is definetely a software issue and not a hardware problem.It seems they fixed one set of dropped calls to create a different set of dropped call issues.They seem to have created a conflict between to seperate peices of software programs,which interferes with the phone keeping signal.Well I hope this helps someone else out,and I will update once a week to let everyone know how it is going with this possible solution.

My phone started dropping calls in my house about a month ago. How can I fix this?

My wife's 4 and my daughters 5C did the same. I called customer service and was essentially told "too bad, if it's in your house we can't really help that". My wife was able to call *228 and her phone now gets 3 bars which isn't good, but its better than not being able to use her phone. My daughter and I can't use *228. We have powered off and back on, removed SIM cards, what else can we do to actually get phone service on 2 of 3 phones in our own home? The phones also drop calls around the corner from our house so it's not just in our home. The phones all work well at school and our jobs. We live in zip code 80525. Help please!

I was not given a ticket number when I called customer service. I was only told the same thing you typed.... "We are unable to guarantee indoor service due to multiple factors that can impact wireless signals with structural interference ranking high on the list; However, other customers are taking advantage of Network Extenders to improve indoor service". I am starting to feel like this is a ploy to get your customers to spend more money. Most reliable network and we have no coverage in a suburban neighborhood??? Hmmmmmm.
As I said in my original post it is not just dropping calls in our home. It is around the corner from our house and in several other areas of the neighborhood, almost clustered to the south end of our neighborhood with the worst being in our home. We have lived here almost a year and had no problems until a few weeks ago. None of our phones have difficulty with calls dropping anywhere else but our neighborhood. It also effects facetime and my wife being able to send picture messages on her iphone 4.
My brother in law has a Samsung Galaxy--on our plan--and came over last night. He said as soon as he gets in our neighborhood his phone varies greatly between 1x and 4G coverage and likes to drop calls.
The only solution proposed by customer service was to shell out $200.00 for a network extender. Again this is not JUST in our home, though it is worst in our home. This is obviously a recent network problem and should not incur a cost on our part to fix it, if we can't make a call until we leave the entire subdivision.
We will be asking people with other carriers to try their phones in our neighborhood. If they have better coverage, our family will be changing carriers. We have been loyal Verizon customers since 1999. Our loyalty ends when we shell out $300 + a month for phones we are unable to even use in our home or our neighborhood. Hoping for a real answer from customer service soon. This generic "we want you to be satisfied and have a great experience" response followed by "Spend more money with us to get coverage that you have had for a year but suddenly don't have" is unacceptable.

MicroCell Dropped Calls

If by failing you mean the light panel going dark and then coming back that is either related to sporadic power outages, either house wiring from the outlet, a failing ac adapter, or the MicroCell itself is having issues with its power board. Even during an update by AT&T, which happens on occassion and usually late at night, the power will blink out momentarily and then the unit will reset itself if that's required for the update to take.
I wouldn't place the MicroCell in direct sunlight. It does generate some heat and the added extra heat from an external source is not good for it overtime. They should have plenty of ventilation around them. Mine sits about 6" away from a North facing window so it gets light in the summertime but not direct sunlight. In fact, all of my MicroCells have been in the same location.
If by failing you mean losing connection to your phones I sill say it has something to do with your setup. Did you see my post about the pfSense configurations?

The Problem is dropped calls using ATT MicroCell. Sometimes we look at the MicroCell when we experience a problem and all or some of the lights are off, then slowly coming back to green. But that is not always the case, sometimes calls drop and the MicroCell shows no changes in the lights. Without the MicroCell we get a signal range from 0 to 4 bars on our iPhones (4s and 5s) that varies by time of day and location in the house. Neither phone is more or less impacted and we drop every other call or more when that call is longer than about 5 minutes. The MicroCell is the older type with the antenna connector (which is not used). History: - Installed MicroCell in 2012, At time time we had UVerse and it was connected directly to the UVerse device. ATT indicated that our local cell tower was getting overloaded during rush hour (it is located near a major highway) and reducing power in order to shed users. Since we live some distance away we would just lose our service, it seems we had no other tower to go to. The situation was elevated to a special case of some sort, a bunch of network analysis was run over a few weeks, and then we were essentially given a MicroCell. The MicroCell always dropped calls, no matter where it was located, but it was better than no service at all during rush hour and I assumed that it was the bad UVerse service. - In 2014, after experiencing a lot of problems with UVerse reliability, we switched to Comcast, which was 20% cheaper and 5 to 8 times the speed and much lower packet loss. As part of this change we installed our own Zoom modem 5341J and a pfSense certified firewall (dumped our older SonicWall). We reconnected the MicroCell and got about the same performance with respect to dropped calls. Today: - Recently, after reading a bunch of posts here, I relocated the MicroCell from the basement upstairs and near a window in an attempt to reduce the dropped calls. Has not helped. We've rotated the MicroCell 90 degrees, and moved it around to no avail. The MicroCell is currently hardwired wired as MicroCell -> HP Procurve 1810 Switch -> Netgate pfSense Certified Firewall -> Zoom Modem. WiFi endpoint is currently about 6 feet away (but was more than 20 ft away before the MicroCell was moved). Our house was built for us in 2011 and contains all new CAT 6 wiring which the MicroCell uses. We have changed the wiring runs, switch ports, etc. to no avail. We have 59.45 Mbps upload, 5.98 Mbps download, 45 ms latency, 7 ms jitter according to www.att.com/speedtest on Sunday morning Aug 9, 2015 at 8:58 CST. For the speed test the laptop was hardwired to the HP Switch. This seems consistent with our firewall logs that shows WAN latency of 9 ms to the first hop and no packet loss around the same time. Prior to Saturday our WAN latency was around 2 ms, but seems to have jumped up over the weekend. I can provide full quality and traffic graphs going back to December 2014. To date we have not been able to correlate dropped calls to local traffic or quality anomalies. We do not show any consistent packet loss to the first hop. In the last 24 hours packet loss had a maximum of 0.4%, average of 0.0%, and minimum of 0.0% and in the last 8 hours 0.0% min, max, average. Our switch shows zero errors on our LAN. - The Question? How do I make our cell phone service more reliable? Having previously designed and managed network services and being an amateur radio operator I have some fundamental understanding of potential problems. Which probably makes me the worst type of customer to have to support, since it means I am not very accepting of the standard, “Do you have it plugged in?” questions. What I lack is a good understanding of the MicroCell and what in detail is happening when our calls drop, even after reading the on-line MicroCell guide by OttoPylot and FAQs by ATT. I do not have my firewall inbound ports open, I have no problem opening them if I was given a domain where requests should come from, but the docs (ATT Firewall recommendations) just say open’em up baby and thats not going to happen. The docs don’t give any reason for having NTP port inbound open. In this day and age, and in this security environment just opening up 5 inbound ports from any source around the globe seems less than professional. Plus I get a pretty consistent but low number of random blocked requests inbound to the firewall (someone is searching for something). It makes no sense, based on what I know today, that a MicroCell using a secure tunnel would need inbound ports open. All inbound communications should use the secure tunnel, but I may not know the complete story. The other difference is that the interfaces on the firewall are set to automatic MTU size. Both sides of the firewall generally show MTU of 1500. I’ve tried 1492 and it did not seem to make a difference so they are set back to automatic. Now if someone has some technical information that explains why any of the above should be changed, then I would be happy to evaluate it. But the problem does not seem to be configuration, since it works some of the time. It appears that the problem is reliability. My firewall and switch are commercial quality, the MicroCell and cable modem are consumer quality. How do I figure out a solution without spending hours on the phone with a customer support person that thinks my IQ is 5? Now that all of our cell phone contracts are done in September, this reliability problem will feature significantly in our decision to continue with ATT after 25 years.

Dropped calls since 2.1 update

What the heck is up with the 2.1 update? It's turned my 3G iphone into a piece of junk. I'm seriously considering breaking my contract because it's almost unusable for it's main purpose: to be a phone. Before the 2.1 update, I had some dropped calls, but not what I'd call excessive. Now, 3 out of 4 calls drop. I think the other 25% didn't drop because they were short, had they been longer they have have dropped.
There are other complaints I have too that have affected the phone since 2.1, constant lockups and resetting, crashing of safari, etc... some things got a little better. However, this is supposed to be a PHONE, so it would be nice if I didn't have to track down a landline to make a call because this POS is falling short.
You do a search on the internet and all you see are headlines about how great the 2.1 update helped call quality and fixed dropped calls. Seems like a bunch of propaganda BS to me. Are others out there experiencing the same problems?

I think I figured something out with these dropped calls. I have noticed that since the 2.1 update my phone no longer fetches e-mail when the phone is sleeping. It get it (15minute interval) when I'm doing anything with the phone but if it's sleeping it doesn't check. Often times, just a minute or two after unlocking my phone say to play a game or look something up in my contacts, it'll chime with about 2 - 3 hours worth of e-mails that it didn't fetch while it was sleeping.
A friend noticed that since the 2.1 update, every time he gets an e-mail while on the phone he drops the call. In his case, he uses exchange so his phone only fetches the e-mail when he gets a push notification. Also I did notice that many times when I drop a call right afterwards my phone chimes that an e-mail came in. So I had an idea that maybe it's dropping the calls when it checks for e-mail. Which makes sense that all of my long calls drop as it's going to check for e-mail every 15 minutes. And most of my incoming calls drop which makes sense that if the phone hasn't been checking and gets "woken up" by an incoming call it's going to check right away. So many of my incoming calls drop right in the first few seconds.
Of course the way to test that is to set my e-mail to manually check only. This really really blows, I only remember to fetch my e-mail a couple of times a day and I have so many that I don't want to deal with them so they pile up on me. However, I haven't dropped a single call since. It's been about a week now and not a single dropped call.
I wasn't able to find any references to this behaviour. Anyone else hear about this? How do we make apple aware of this??? Obviously it's a bandaid - I've gotten call reliability back at the expense of practically throwing my e-mail out the window but at least I'm not going to chuck the phone out the window now.
Well I might for internet access... Safari crashes are driving me nuts too...

Rezound Dropping Calls

Okay, I've had the HTC Rezound for about 10 months now and have had pretty smooth sailing with it. However, lately there's been a huge problems. Sporadically over the past few months my phone has been dropping calls. I'll be chatting with someone for about 20-30 minutes and all of a sudden, without me moving, the person will start to break up and then BEEP the call is dropped. Previously, a battery/sim pull would improve things, however today it has not. I've tried switching it from 3G/4G to just 3G, I've reseated the battery and SIM, no good. I really am trying to avoid a reset, just tried calling Tech Support but they are in the middle of a computer update and can't help me right now. My phone shows 3G and full bars and is having no problem with data connectivity. Also I live in the middle of town so I'm not in a weak signal area at all. Anyone have any thoughts?
Thank you!
Casey

Yes, this has been a recent problem for me as well here in Los Angeles. I have had my Rezound for 4 months and upgraded to ICS in August.
I can no longer keep a call without it dropping in my apartment. Lately from my apartment my phone has 2 signal bars before a call, then drops to 1 bar during the call, then zero and drops out completely. This has been happening since September and I have never had this problem before. I thought I was the only customer experiencing this. Either this is a ICS issue or the phone's antenna is flawed.

Immediately dropped calls

Yesterday, whenever I made a call, the call would be dropped (with "call failed" on the screen) within 3 seconds. Very irritating.
I found that if I turned the phone off, then back on again the problem went away.
Do the Apple folks trawl this forum looking for ways to improve the iPhone?
Message was edited by: Peter Ritchie, spelling corrections

I've had this same problem with dropped calls. I live in the Seattle area however. It really got much worse after the 2.1 update. I would get at least 4-5 dropped calls a day.
While it is true that that turning off 3G and switching over to EDGE will get rid of the dropped calls, this is really a temporary solution. I mean, it is the 3G iPhone after all so no sense in using it on "crippled" settings.
Here is the chronology of what I did and the solution I've come to. I think I have found something that works but it is a little early still (so far NO dropped calls and no calls going straight to VM for 3-4 days)
I first rebooted my iPhone...still dropped calls.
I then reset my network settings....still dropped calls.
Then I switched out SIM cards....still dropped calls.
Then I did a RESTORE but NOT from backup. I knew this would be a little painful but I was willing to try anything. Now just know, if you don't sync your iPhone with Outlook or MobileMe or something, you will lose all your contacts and appointments. I would recommend actually doing a quick sync before you do a full restore in case you wanna transfer any new info from the iPhone back to your computer including Apps and iTunes purchases. Just know that regardless, you apps will all reset to there initial settings. This is kinda a pain (you'll lose game saves, apps preferences, etc...)
The restore process will take me about 20-30 minutes to reinstall the software and everything (not sure but I've been told could take up to 45 minutes)...just be patient (do it before you go take a shower or times when you don't think you'll need your phone). Afterwards, you may notice your iTunes content by default does not transfer over but your apps should. You'll have to go into the iPhones settings on iTunes and check off what you want to sync over.
So far so good....No dropped calls. I suspect somewhere along the line...the firmware or OS became unstable or corrupted but doing a full restore fixed it. I am not sure a restore from backup works and you could certainly try that first but it sounds like it does not work.
Good luck.
Arman...

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"

My IP4 drops calls alot call failure is what it says on screen even in good service area and good bars on phone

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