Loss of sound to caller / Also Shared Line Appearances

1. We have had on several occurrences a situation where we (people in the office using the 941 phones) can hear the caller (or person we call outgoing), but they are unable to hear any sound from us. Rebooting the SPA400 has seemed to solve the problem. The problem has happened at least 5 times today, each time requiring a reset of the SPA400 before the external side of the party is able to hear us.
2. All of our 941's have their "prviate extension" configured on the first line button, and a Shared extension configured on the remaining 3 buttons. When I first setup the system, after a call was answered on the shared extension, the other phones would continue to show the light blinking red until the caller who answered the call disconnects. Something has changed in our system where now once the call is answered, the other "shared" phones immediately go back to a green light. Any idea what I might have changed to cause this change in behavior of the system?
Thanks in advance.
Nick
Our configuration is SPA9000, SPA400 (3 PTSN lines), with SPA941 phones. We are only using PTSN lines at this point.

Similar Messages

Shared line appearance service scenario

I have two 962 phones with version 6.1.3(a) for SLA test. Each has number 3000 and 5000.
configuration summary below:
            3000       5000
EXT1    3000     5000
Phone
Line1   ext 1, shared, 3000                       ext 1, shared, 5000
Line 2 ext 1, private, 3000                   ext 1, private, 5000
Line 3 ext Disabled, shared, sip5000       ext Disabled, shared, sip3000
   fnc=blf+sd+cp;sub=sip5000@$PROXY;ext=5000@$PROXY
                                                             fnc=blf+sd+cp;sub=sip3000@$PROXY;ext=3000@$PROXY
   Line 3 ext Disabled, shared, sip5000       ext Disabled, shared, sip3000
   fnc=blf+sd+cp;sub=sip5000@$PROXY;ext=5000@$PROXY
                                                             fnc=blf+sd+cp;sub=sip3000@$PROXY;ext=3000@$PROXY
As a result of setup using Setup Wizard program verion 2.1
At 3000 phones
EXT1
     Shared Ext:"shared" Shared User ID: 3000
   Proxy; x.x.x.x
     Subscriber Information: Display Name: sip3000, User ID: 3000
Phone
   Line Key 1
Extension : "1",
            Short Name : "3000"
Shared Call Appearance: "shared"
   Line Key 2
Extension : "1",
            Short Name : "3000"
Shared Call Appearance: "private"
   Line Key 3
Extension: "Disabled"
Short Name : "sip5000"
Shared Call Appearance: " fnc=blf+sd+cp;sub=sip5000@$PROXY;ext=5000@$PROXY
Line Key 4
Extension: "Disabled"
Short Name : "sip5000"
Shared Call Appearance: " fnc=blf+sd+cp;sub=sip5000@$PROXY;ext=5000@$PROXY
At the phone's screen, it displays like that a first line has folded phone icon, second has normal phone icon, the third and the forth line has 3 by 4 keypad icon.
At 5000 phones
Same with above but changed the number 3000 to 5000 and visa versa.
The configuration show above is the results of setup wizard, not manually configured.
The "System Configuration Using the SPA9000 setup Wizard, verion 2.1, Installation and Configuration Guide" says that
"In this example, a station is configured with the name Sales2. Line 2 is
configured with a Shared Line Appearance for a station named Sales1. After
these settings are submitted, the button will display the name of the selected
station, Sales1. When an incoming call is directed to Sales1, both phones will
ring, and Line 2 on Sales2 will flash red. The user at Sales1 can answer the
call normally, or the user at Sales2 can answer the call by pressing the Line 2
button. If the call is answered and put on hold, it can be resumed from either
Sales1 or Sales2."
<Question 1>
The underlined sentence does not working.
2000 make a call to 3000, both 3000 and 5000 ring, 3000 answered and put this call on hold, and try to retrieve at 5000 by push the line displayed "sip3000". but It make a ring on 3000 with new line !!!!!!!!!!!
<Question 2>
The difference what was written in this potsed threads is that the Line key3's extension is not "Disabled" but another EXT number. (for example EXT1, or )
For example, At station 3000 -> phone -> like key 3, when I changed the Extension "Disabled" to ext "1", If I make a call to 5000,this phone 3000 does not ring simultaneously.
<Question 3>
I added EXT 2 for each phone's subscriber information.
At Phone 5000,
EXT2
     Shared Ext:"shared" Shared User ID: 3000
   Proxy; x.x.x.x
     Subscriber Information: Display Name: sip3000, User ID: 3000
            3000       5000
EXT1    3000     5000
EXT2      5000     3000
Phone
Line1   ext 1, shared, 3000                       ext 1, shared, 5000
Line 2 ext 1, private, 3000                   ext 1, private, 5000
Line 3 ext Disabled, shared, sip5000       ext Disabled, shared, sip3000
   fnc=blf+sd+cp;sub=sip5000@$PROXY;ext=5000@$PROXY
                                                             fnc=blf+sd+cp;sub=sip3000@$PROXY;ext=3000@$PROXY
   Line 3 ext Disabled, shared, sip5000       ext Disabled, shared, sip3000
   fnc=blf+sd+cp;sub=sip5000@$PROXY;ext=5000@$PROXY
                                                             fnc=blf+sd+cp;sub=sip3000@$PROXY;ext=3000@$PROXY
The shared line icons are changed as "folded phone" icon.
PBX Status shown below:
Registration Station User ID IP Address Reg Expires(s) User-Agent
sip3000 3000 192.168.10.2 37 Linksys/SPA962-6.1.3(a)
sip3000 5000 192.168.10.2 59 Linksys/SPA962-6.1.3(a)
sip5000 5000 192.168.10.3 45 Linksys/SPA962-6.1.3(a)
sip5000 3000 192.168.10.3 79 Linksys/SPA962-6.1.3(a)
sip4000 4000 192.168.10.4 45 Linksys/SPA942-6.1.3(a)
then, "If the call is answered and put on hold, it can be resumed from either
Sales1 or Sales2." ==> It works !!!!!!!
Does a phone should register with not only its own number but also shared line phone's number ?
Does it correct configuration for Shared line appearance ???
<Question 4>
2000 call to 3000, 3000 answer, 2000 and 3000 is connected.
3000's line 1's LED is red and icon is "3000 <-->".
5000's shared line 3 's LED is red and blink intermittently and icon is "sip3000 <-->".
If I push the shared line 3(sip3000) at 5000. call was picked up, thus 2000 and 5000 was connected.
BUT, At 3000 there is no lines are turn RED (it means that the shared line is not monitored).
I think the shared line "sip5000" at 3000 should be turn RED.
Is it normal operation ?
<Question 5>
and finally, what scenarios does spa9000 supports with related to SLA?
A --> B(C), C(B) ring simultaneously.
hold, resume by each other.
transfer or conference
pickup established user by push shared line.
what else ? If is there any service scenario document please let me know.
what about intercom?
When sceretary answer a call, how to let boss know? just seize a new line, make a call and transfer after answer ?
or, assign new line with func like intercom and push that line to notice a new call to boss and then
transfer or hold by sceretary resume by boss etc.,
Message Edited by nature21 on 03-18-2009 11:09 PM

as of now its hard to tell the exact problem with your setup. i suggest that you try to reconfigure the device again. you may also try to check the admin guide for the device.

Shared line appearance for FXS phones on SPA9000

Hello again!
(I hope asking these different questions in separate threads makes it easier.. not harder.)
If I'm using shared line appearance, can my FXS (regular cordless phones) pick up lines on hold? Can they put a line on hold? Is it easy?

I've never read or seen anything about SLA on the FXS ports.. but I guess you can always just try and see what happens. Configure one FXS port with the same userid as the shared line and see what happens if you put a call on hold. Since hook flash is used to put a call on hold / take it back, I figure if it worked, all it takes would be going off-hook and you'd have the call, and do a hook flash to put it back on hold.
There are two essential pieces to tracking down a problem with your VoIP equipment:
The configuration of every device involved
SIP protocol traces
And don't forget: there's no such thing as giving too much information when describing a problem.

Loss of sound during calls following updating iphone 4 with IOS5

I've recently updated my Iphone 4 with the IOS5 update. Since then I've had terrible problems with loss of sound on phonecalls. I can hear others but they cant hear me - loss of sound is patchy during a call. I've cleaned mic with a soft brush and rebooted but problem still exists. Any ideas before I restore factory settings/ and or take back to the shop?

I have a Iphone 4 and when i try to make calls, i cannot hear anything unless i put it on speaker, and sometimes if i try to turn the call volume up during a call without the speaker it almost sounds like the speaker is blown out. I just bought this phone online 2 days ago so i don't understand how that could happen.
Is anyone having this problem? I'm am totally lost with what to do other than return the phone in and see if i can get another one?
Please help!!!!!

Can sources be filtered calling a shared line?

We have multiple sites, one of which houses a Helpdesk with a shared line.... (say, "Site A"). One of the members of the Helpdesk sits at a different site from the rest of the group ("Site B"). Is there a way we can set it up so that the person at Site B can receive calls from all users (Sites C, D, E, etc.) except those calls coming in from Site A (using the same shared line)? I know we can set up filters but I'm only seeing instructions on how to filter users dialing a number as opposed to what number they're dialing from.
Thanks in advance.

there are no inbound DNIS filters to send calls to certain destinations in CCM by default. (upto 4.13 that i know of)
you may need to look into an AA or IPCC solution. something that can determine how to route the call based on DNIS/ANI information.

UCCX and Shared Line appearances

If an acd group also has a shared line ( not the acd line dn ) on each of the acd sets, will that affect the agents' wrap up timer, and if so which bug report is that documented on?

1. We have had on several occurrences a situation where we (people in the office using the 941 phones) can hear the caller (or person we call outgoing), but they are unable to hear any sound from us. Rebooting the SPA400 has seemed to solve the problem. The problem has happened at least 5 times today, each time requiring a reset of the SPA400 before the external side of the party is able to hear us.
---try setting VAD to ON , set preffered coder to g711u on the spa400, don't forget the magic power cycle after saving settings.

Shared line LCD issue

Customer is running CCM 4.2(3) and has 7960s with 7914s setup with shared line appearances to monitor the other users. When a shared line is ringing on the 7914, the 7960 shows the call info on the LCD of the 7960. The receptionists don't want to see that. They only want the status to show on the 7914. I've tried the Always Use Prime Line setting, but didn't work.
I know its just a setting somewhere I'm missing.
Any thoughts?
Thanks,
Kevin

Under 4.2(x) this BLF is (only) used for the new "Directed Call Park" feature, and cannot be used for "monitoring" a shared line. As far as I know, this is part of the 5.x releases (together with a Presence Server?).
Trying to suppress (or minimise) the displayed info on the receptionists , you could try to change the "Line Settings for this Device", "Ring Setting (Phone Idle) & (Phone Active)" to disable, and deselect all options under the "Forwarded Call Information Display".
Also leave (set) the "Auto Line Select" to "Disabled", otherwise they risc the possibility to pickup accidently a "shared line".
Succes.

Hunt Group Shared Line question

Customer has the following call routing scenario:
CCM 4.2(3)
CTI Route Point is CFwdAll to a Hunt Pilot.
Hunt Pilot contains Hunt List containing a broadcast Hunt Group of 5 phones.
4 of these 5 phones each have the other phones' shared line appearances (ring disabled) so they can see if they are on the phone.
Issue - When incoming call hits the CTI RP, it then goes to the Hunt Pilot, Hunt List, then it "should" ring all the phones in the hunt group. But it only rings the very first DN in the Hunt Group and the DN of the phone that is not setup as a shared line on the other phones. The 4 phones that have each other's DNs setup as shared lines only see the call info popup on the LCD screen showing that the first DN is receiving the call, but their phone doesn't ring.
As a test, I switched the order of the hunt group DNs. After the switch, I realized that only the first number in the broadcast hunt group will ring. Kind of defeats the purpose of a broadcast hunt group. :(
I am starting to see where the issue may come from. If a phone has shared lines setup for DNs that are in a broadcast hunt group, each time a call comes in, then the phones would fill up the LCD with the call info for the 4 shared line DNs.
Any suggestions on allowing phones to be in a broadcast hunt group, and having shared line appearances to monitor the status of the other phones in the hunt group?
Thanks,
Kevin

Hi Kevin,
You are right on the money! Here is an odd tech note that applies to CCM 4.1.X , 4.2.X and 5.0.x. The reason I say odd is that it is hard to read or make sense of, but it is saying exactly what you have discovered.
Note: Do not put DNs that are shared lines in a line group that uses the Broadcast distribution algorithm. Cisco CallManager cannot display all DNs that are shared lines on devices where the DNs are configured as shared lines if the DNs are members of a line group that uses the Broadcast distribution algorithm.
From this doc;
Cisco CallManager Administration Guide, Release 4.1(3) Line Group Configuration
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803ed665.html#wp1036009
In the current environment I can't think of a good way to accomplish your goal of being able to monitor each others activity while using Broadcast hunting.
I know this doesn't help, but at least you know why it works the way it does. (and have some documentation)
Take care!
Rob

Shared line sometimes does not ring on some phones

I'm having a strange issue that appeared a couple of weeks ago. I have not made any upgrades since October so I don't think that's the issue. I'm running CUCM 8.5.1.15900-4.
The issue is that one of my phone's shared line appearance does not ring while the others are ringing. It seems like that phone is omitted from the broadcast sometimes. This doesn't happen all the time. Most of the time, all the phones with the shared line appearance will ring. I'm not sure if it is only happening on this phone. It is very hard to reproduce.
Has anyone had this issue? What are some debugs I can collect to show what physical phones CUCM is ringing when the calls are coming in. I want to confirm that it is omitting that particular phone.
Vince

Hi Jeff,
I never went as far as to open a TAC case.
Most things I tried have corrected the issue temporarily (resetting the DN in CUCM, resetting the phones, upgrading firmware). I acutally upgraded CUCM from 8.6 to 9.1.2 just under a year ago and haven't had the "ringing" issue on the shared line since then. Not entirely sure if that was the reason though.
Since that upgrade I have other issues with this same shared line, most notably held calls not showing up on all the phones at intermittent times. 9 times out of 10 resetting the DN corrects this for about 6 months. That other 1 time, a reset via CUCM doesn't actually reset the phone and I either have to physically unplug power from the phone or do a shut/no shut on the switchport (effectively power cycling the phone).
I've created a job under bulk administration that resets the phones once a week (during closed hours). My thinking was these phones are like computers and need to be rebooted every now and again....

88xx Merge/Barge Notification in Call Session bubble across shared lines is not consistent.

Hi All,
Here is the setup and the steps I used to see this issue…
Phone A - 601052 is a single line phone
Phone B - 601053 is a single line phone
Phone C - 601051 has shared lines with 601052 and 601053
In this case Phone A (601052) calls Phone B (601053). All is normal and working great.
Then on Phone C (With All Calls On and Use Primary Line On) we see everything looks normal.
Then on Phone C hit the Session Key, We get the Prompt for Merge, say Yes, and then we get Unknown on 601052 in the call session and we believe it should say conference.
If I hit the 601052 shared line PLK on (601051) I just get UNKNOWN w/o the On 601052 and if I hit 601053 shared line PLK the Call Session ID say "Conference".
The line that dialed out 601052 (which called) 601053 shows the original dialed call. Shouldn't it say "Conference" too?
Please let me know the exact behavior.
Thanks,
Guru

Please anyone help me

No sound in my iphone 4s. I can make calls but no ring tone or lock sound and siri also doent talk. My sound settings are ok and it is not in silet mode. Please help

no sound in my iphone 4s. I can make calls but no ring tone or lock sound and siri also doesnt talk. My sound settings are ok and it is not in silent mode. Please help

Reset, restore, stop trolling.

Spa3102 would not forward a voip call to pstn line

Good morning.
I've done the implementation provided here http://community.linksys.com/t5/VoIP-Adapters/SPA-3102-and-softphone-to-
make-calls-via-pstn-line/td-p/326390 .
It is a way to use for outgoing calls a given pstn line from anywhere I have internet (voip to pstn).
The spa3102 is connected to a router (with an active DHCP server and ip 192.168.1.1) from where it takes the internal
ip (192.168.1.3).On the same network is also a computer , connected to the router ( with ip 192.168.1.2). The spa3102
is set to bridge mode and thus inactivates the function of the router (on SPA3102), and it functions as a simple
network device . I have done port forwarding (from the router) to 192.168.1.3 (SPA3102) for the port 5061 (PSTN
LINE) ( but for 5060 for the LINE 1 also). I want to make calls from a voip softphone (x-lite 4) to the SPA 3102 and
this to forward the voip calls to PSTN line to which it is connected. In x-lite the SPA3102 is set as a proxy so that
i can type the phone number I want to call without being followed by the SPA3102's ip each time ( eg on x-lite I
give call number 2101111111 instead of 2101111111 @ wanip: 5061 where wanip is the external ip of the router).
When x-lite is running on the computer that is on the same network with the SPA3102 everything works as expected. A
voip call is made from x-lite ( using as a proxy the wanip everytime, or even for test purposes the dyndns domain
that i set up for this reason), this call is passese PSTN line and the phone of the called party rings . At x-lite
COMES indication "call established ".
The problem occurs when I do the same procedure from x-lite installed on a computer belonging to another network (
e.g. in another building with its own internet connection , own router, own computer , etc. ) . Always using the
wanip the x-lite makes the voip call to the SPA3102, writes "call established" ( meaning it connected to SPA3102) but
never routed the call to the called party ( the SPA3102 did not forward voip calls it receives to the PSTN line ) .
Trying to find what 's wrong I've tried to disable all firewalls (soft and hard from all involved machines ) . The
behavior is the same either the computer that makes the successful calls is connected to the network directly to the
router or through the port "ethernet" on the SPA3102.
What is the difference in these two voip calls to the SPA3102 and the one " triggers " it to forward the call to
PSTN line and the other does not ?
Thanks now for any ideas you give .

The audio sound problem is more than likely also associated with the overall addressing problem initially encountered. As you may know, using the sip protocol the sip signalling exchanges ip addresses to be used for both the sip signalling and the exchange of rtp sound packets. In addition there is an exchange of port numbers to be used for the exchange of rtp sound packets. The sound is exchanged by two separate streams of packets, one stream in each direction. The result is an ip address and port number for the rtp packets flowing from the SPA3102 to the softphone and a different ip address and port number for the rtp packets flowing from the softphone to the SPA3102.
In your previous posting you mentioned that you "set the minimum EXTernal rtp port at the sip tab". Changing the "EXT RTP Port Min:" is an unusual change to make and in my opinion would only be made in special circumstances. Actually, I ran some tests and I'm not sure exactly what that setting does. In my tests it didn't appear to affect the rtp port number used in a predictable manner.
The common changes to make for audio problems typically would be to setup a STUN server. A STUN server is an external server that echos back to the initial sender the external ip address and port number that the STUN server received with the message received by the server. This allows the sender (SPA3102 or softphone) to determine its external ip address and external port numbers for both the sip signalling and rtp packets.
A STUN server is commonly recommended to be setup with the following settings in the SPA3102:
PSTN Line Tab:
NAT Mapping Enable: Yes
Sip Tab:
Handle VIA received: yes
Handle VIA rport: yes
Insert VIA received: yes
Insert VIA rport: yes
Substitute VIA Addr: yes
Send Resp To Src Port: yes
STUN Enable: yes
STUN Server:
The following web page has a list of "Public STUN Servers"
http://www.voip-info.org/wiki/view/STUN
You are using CounterPath's XLite softphone. stun.counterpath.net is a STUN server on the list.
I see XLite also has a setting to use a STUN server on the "Topology" tab.

Are there any billing benefits to have a family shared line?

Are there any billing benefits to have a family shared line? Currently I have a family shared 700 minutes w/ 500 texts. Would it be just as cheap to have to separate lines? I am asking this because I have my bro on my plan and am asking if it is saving me any money . Also why do I have to keep dialing *228 to decent internet connection on my phone. so any time I go anywhere i need to dial x228 to get 3g?

jackscagnetti wrote:
yes he is splitting it in half.
I only get 15% Access Discount & 20% Feature Discount on my line. Which comes out to -$15
I do not get any employee discount on the second line.
so $60-9+29-6=74 for one line not sure about the second line pricing...
Hi jackscagnetti, and thanks to our forum members for the great input on this monthly pricing question.
Using the numbers included on your post it seems like you have a 69.99 calling plan and a $29.99 data plan. Please see break down of fees below and difference between family share plan and individual calling plans.
Current family share plan:
Primary line
$60.00 - Calling plan
$29.99 - Data feature
$89.99 - Subtotal
$09.00 - Credit 15% calling plan
$06.00 - Credit 20% data feature
$74.99 - Total monthly fee for primary line (tax and surcharge not included)
Secondary line
$09.99 - Calling plan
$29.99 - Data feature (if Smartphone on this line)
$39.98 - Total monthly fee for secondary line (tax and surcharge not included)
$114.97 - Total for 2 lines.
2 individual lines with a $39.99 calling plan as used on previous scenarios in this forum/thread.
$39.99 - Calling plan
$29.99 - Data feature
$69.98 - Subtotal
$06.00 - Credit 15% calling plan
$06.00 - Credit 20% data feature
$57.98 - Total monthly fee for primary line (tax and surcharge not included).
$115.96 - Total if system allows employee discount on both lines (tax and surcharge not included).
$127.96 - Total if system allows discount on one line only (tax and surcharge not included). In both scenarios your monthly fees are lower with a Family Share calling plan.
Please keep in mind that we can only review your employee discount specifics if we access your account information. Feel free to contact us at 1-800-922-0204 at your convenience for further assistance with any billing issues and be prepared to provide account verification information.

Shared line/Transfer error

CCM 4.2; Shared line
A receptionist has told me there are times when handling a call on a shared line, and in process of transferring the call that a second call rings to the line and causes an error message "number is not valid" (or something to that affect)and to "please try again later". At that point the receptionist presses "end call" to disconnect from tranfer mode and she is able to reconnect with the call being transferred. The second call was answered and handled, but the first one wasn't able to be transferred due to ??
I'm told by the other receptionists this has also happened to them.
This shared line is an emergency line, and needs to work smoothly. It's set for maximum of 5 calls, with a busy trigger of 2.
Any thoughts on how to prevent the error message from occuring, and tranfer to be allowed to complete when new call is ringing through?
Thank you, Nat

Hey Nat,
Good stuff! Here is some info for "On Hook" transfer;
Onhook Call Transfer
Modifications to the Call Transfer feature add the onhook (hangup) action as a possible last step to complete a call transfer. The Transfer On-hook Enabled service parameter, which enables onhook call transfer, must be set to True for onhook call transfer to succeed. If the service parameter is set to False, the onhook action ends the secondary call to the third party.
In the existing implementation, if user B has an active call on a particular line (from user A) and user B has not reached the maximum number of calls on this line, the Cisco IP Phone provides a Transfer softkey to user B. If user B presses the Transfer softkey (or Transfer button, if available) once, user B receives dial tone and can make a secondary call: user B dials the number of a third party (user C). Cisco CallManager provides a Transfer softkey to user B again. If user B presses the Transfer softkey again (or Transfer button, if available), the transfer operation completes.
With the new onhook call transfer implementation, user B can hang up after dialing user C's number, and the transfer completes. Both the existing and new implementations work in both the case of a blind transfer (user B disconnects before user C answers) and also in the case of a consult transfer (user B waits for user C to answer and announces the call from user A).
The previous implementation remains unchanged: user B can press the Transfer softkey twice to complete the transfer
From this doc;
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a008055cd66.html#wp1088627
When you say "It's after dialing the extension line that they get the error of "not a valid number, please try again later"." Is this a visual message on the display (Probably) or an audible message?
Let me know
Rob

Lost sound on calls

I've had this phone for about a month or so and besides the size of it, it's awesome.
However, in the last few days, I've lost all sound on calls. Putting the speakerphone on allows me to hear things going on in the call, so I know it's connecting (I can also see the phone I'm calling ring). At this point, I've done a factory reset and removed the last few applications I've installed so I'm kind of at a loss why the earpiece just up and stopped working.

Hello nonades! Sorry to hear of the trouble you have been having with your phone. You sound pretty familiar with it from the troubleshooting you have done so far! I want to check into this further for you. I have sent you a private message here in the forum. Please check your inbox and reply to my private message so we can get started. Thanks in advance!
bryans_vzw
Follow us at Twitter @vzwsupport

Loss of sound to caller / Also Shared Line Appearances

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