SIP trunk INCOMING/OUTGOING calling issue

Dears,
i am facing issue i.e inbound / outbound calls are not wokring after configuring SIP trunk . the call flow is
SERVICEPROVIDER--------sip trunk------VG --------------sip trunk ----------CUCM
For Outbound calls ---error-----tot tot tot
For Inbound Calls -----error--- Silence
Below are the configuration and debugs:

Appreciate if some provide me Sip configuration and troubleshooting guide...Thankyou

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To me this sounds like the RTP stream problem - it probably does not pass the firewall / nat on the way between SIP provider and you.
What may have happenned is, that either on your Belkin router/NAT, or on internet ISP's router/NAT the SIP ALG (Application Layer Gateway) was enabled.
If this is the case, you need to disable SIP ALG on your router and/or ask your internet ISP to disable it on the firewall router.
Another issue may have been if you have (accidentaly) changed the port forwarding on your Belkin router, and/or the PAP2T local LAN IP address changed and the port forwarding no longer works.
So my proposals to check would be :
- check and DISABLE SIP ALG on your Belkin (if router has this feature)
- check the port forwarding on your Belkin router ... IF you setup port forwarding for 5060/61 you MUST set the same for RTP ports (16384-16483)
- if you don't have port forwarding on your Belkin, try to setup forward of 5060-5061 AND 16384-16483*.
*(If this is too many ports, set the the RTP portrange on PAP2T to 16384-16394 for example and then forward only that range)
- if neither of above helps, change your Line1 SIP port from 5060 to let's say 6070, and RTP ports to the range let's say 17300 - 17310
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Under one contact name I have 4 to 5 Numbers (Local & International).
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Over the last month, Skype has been giving me and others in my house issues.List of symptoms:Incoming/Outgoing calls do not ring. However, you recieve a missed call notification.Incoming/Outgoing calls ring but are immediately dropped.This has happened to me and others in my house when using a laptop with wifi and/or a PC with an ethernet connection. Sometimes it works, sometimes it doesn't. I have the most current version on Skype (in fact, installed an update last night). I do have comcast but figured it wasn't an Internet issue since I can access all web sites. This only seems to happen when using Skype. Anyone else having this issue? Know of a way to fix it? Thanks.

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http://www.apple.com/support/manuals/iphone/
Hope this helps out. Oh and if you want, you can right click on that file and choose to download it to your system. Otherwise it will probably display right in your browser.
Message was edited by: QuickDraw

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Im not contracted to BT but go via TalkTalk, and just want to find out if I am setting the DECT phone up incorrectly!

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I wish to keep my incomming / outgoing calls switched off, while 3g / E should be enabled,

I wish to keep my incomming / outgoing calls switched off, while 3g / E should be enabled, I dont want to be disturbed with incomming calls while surfing, is there some way to do this in Iphone 4.?

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SIP trunk incoming and outgoing calls issue

Hi Everyone,
We recently installad a SIP trunk and terminated on CUBE and CUCM but we have issues on incoming and outgoing calls, When someone dial in from outside he keeps listening the dailing ring even after we pick up the phone and at the end the callers time exipres and call gets disconnected.
For Dailing out, the dialed number rings and caller hear the dailing ring as well but if someone pick the phone it apprears that call is connected but no audio in it, dead air.
Our call flow is as
IP Phones => CUCM --->SIPTRUNK--->CUBE=>SIPTRUNK=>SP
I have attached the config for CUBE and debug ccsip messages output for both incoming and outgoing calls.
Please if some help in sorting out this issue, Thanks in Advance
Tasneem

Inbound call>>>>
The reason you are experiencing this is that your CUBE is requesting PRACK and your provider is not responding to it..
Here we have your cube sending 180 ringing with "Require 100rel"..This was sent several times and your ITSP didnt respond probably because they do not support 100rel...(It is Huwaei after all, they do what they like)
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
From: ;tag=sbc080552fph4hp-CC-25
To: ;tag=256F3440-12C
Date: Thu, 16 Jan 2014 13:31:34 GMT
Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
CSeq: 1 INVITE
Require: 100rel
RSeq: 2507
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "TEST STC" ;party=called;screen=yes;privacy=off
Contact:
Record-Route:
Server: Cisco-SIPGateway/IOS-15.2.4.M2
Content-Length: 0
AFter the CUBE didnt get any response, it then replied with Gateway Timeout...
Jan 16 13:31:54.550: //31880/5A4406E48184/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 504 Gateway Timeout
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
From: ;tag=sbc080552fph4hp-CC-25
To: ;tag=256F3440-12C
Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
CSeq: 1 INVITE
Reason: Q.850;cause=102
Content-Length: 0
I suggest you disable this parameter..and test again
voice service voip
sip
rel1xx disable
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Sip outgoing calls issue, getting 403 forbidden sometimes.

Hi!
This is my connection to provider
ISP---Sip---CUBE---Sip---CUCM---Sip---7821
I've got a problem with otgoing calls.
On sucessfull calls my CUBE sent to ISP Invite with SDP, then gets 100 trying, 183 progress sdp, 180 ringing and rtp after it. The call is ok. Hold, transfer, redirect works fine.
But sometimes (maybe 30-40% of outgoing calls) I sent Invite with SDP, then get 100 trying and after it my cube receives 403 forvidden from provider, cause code 57. Session breaks. What's the matter with it? Incoming calls are ok.
Also I disabled providers MOH on this cube by using no ip cef on it and using software mtp on sip trunk fom CUCM.

If the provider is the one sending that to you then it would be best to get with them. The initial INVITEs and TRYINGs look good from what you can tell? Is there any dialplan differences between the phones? Is it the same phones causing this problem or is it random? These are things you need to look into.

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Hi Folks,
I am using a Cisco 2821 as a router that will convert a SIP trunk to an E1 PRI. Si my setup is:
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"Debug CCSIP calls"
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4BB4F194
State of The Call : STATE_DEAD
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Called Number : 909
Source IP Address (Sig ): 172.29.x.xxx
Destn SIP Req Addr:Port : 10.200.7.157:5060
Destn SIP Resp Addr:Port : 10.200.7.157:5060
Destination Name : 10.200.7.157
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 172.29.5.210
Source IP Port (Media): 16786
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 28
Disconnect Cause (SIP) : 484
Can anyone shed some light on the area that I need to focus on? This is my first attempt at SIP and I am confused :)
Thanks.

Hi
909 is what I meant to dial as that is the help desk for the telco.
I tried mobile numbers as well getting the same error codes. And international numbers.
If it's based on the called number being wrong then I guess I will have to play with the calling party ID and call type as well... Maybe this is causing it to fail??

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Welcome to the Support Communities!
The following article will provide some troubleshooting steps for this issue:
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http://support.apple.com/kb/TS1630
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SIP trunk INCOMING/OUTGOING calling issue

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