Routing incoming calls
I've recently starting experimenting with Cisco Call Manager (version 8.6). I've been able to figure out routing outgoing from the one extension I have setup, but I can't seem to figure out how to route incoming calls to that extension. I'm running a Cisco 2901 as an MGCP gateway, and all calls are routed through a VIC2/2FX0. How do I go about routing the incoming calls?
Alright, so in the window, I have
Number Type
Prefix
Strip Digits
Calling Search Space
Use Device Pool CSS
What would I put under each? Sorry, again, I am brand new to this, and I haven't been able to find any good guide documents online on the subject. Basically, what I want to do is route all incoming calls on voice port 0/0/0 to extension 1001. All outgoing calls are functioning normally.
Is there a guide that I missed somewhere that would walk me through this?
Similar Messages
-
Inter-Trunk not route incoming calls from out
Hi,
I setup one extra gateway where I try to route part of our calls. So far I have success to route internal calls into there, but when I'm making a test call from outside that ends into "number is not used" problem.
I have:
- Route ready, elsewere the internal calls are not working.
- PSTN usage, linked to the Route
- Trunk configuration where I have selected the PSTN usage
- Incoming numbers are coming in E164 format
I have also tested the "Test-CsInterTrunkRouting" and that gives "pass":
FirstMatchingRoute : Description=;NumberPattern=^\+358123654789;Name=Test
Gateway;SuppressCallerId=False;AlternateCallerId=
MatchingUsage : Test PSTN Usage
MatchingRoutes : {Description=;NumberPattern=^\+358123654789;Name=Test
Gateway;SuppressCallerId=False;AlternateCallerId=}
But still, when I made a call from outsited the OCSLogger shows that mediation server try to offer call to Front-End which says only: "SIP/2.0 404 Not Found" and then bye-bye.
What is the missing magic, which made the mediation server to see alternative route? I hope it is not required that mediation server must be collocated on the Front Ends, as that one I do not have.
Any good ideas?
ps.
I'm not sure does it matter, but my Lync gives "SIP/2.0 403 Forbidden" when there is coming call from extra gateway. But as the calls into there works, then I don't see why external calls should not also work.
PetriCould it be even so, that intra-trunk routing requires consolidated mediation server? As the call is owned by the Mediation server (stanalone), and it is trying to offer that to FE. FE reply "does not exist". Because of the standalone Mediation
server does not have the call routing engine like FE have, the call is lost.
I started to think above as Lync users are able to call to that number. So FE is able to do the routing and get calls into the correct place.
I have to say also, I have read
Ken's blog about inter-trunk routing, I have to say that I'm not so sure what he means by this: "Fortunately, in most cases, adding PSTN usages to the trunk has no effect, since there is almost always a Lync user assigned to the incoming phone
numbers". Why to add additional routing for the numbers which are already inuse? I hope it is not required, that you need to have a users ID for each number you do the inter-trunk routing?
Petri -
Routing incoming calls from outside
Hello everyone. I am new at Cisco VOIP and i need some help.
I manage to create VoIP on my router C2911 and i have 3 IP phones. I have one voice port that is conected to my country telekom(ground line whose number iz YYYYYY). My phones have 3 digits extensions. Example 222, 333 and 444. They can call each other localy. I configured dial peer pots, and when i make call from my IP phone to my cell phone it goes through voice port, and on my cell phone is shown that the ID of the caller is YYYYYY.
Now i want to configure that when i call from my cell number YYYYYY, router "answers" and ask me what extension do i want and when i type on my cell 222, my IP phone with that extension needs to ring.
Did anyone had same thing to configure?
All the best.Hello
If i understand your question correctly. You will need to enable as AA "auto attendant" to help any incoming call . For example 0 for operator , if yu know the extension press 1. You have two methods :-
1-CUE :is a module (SRE Module) which added to your Cisco router . This module is HW should be purchased by your account manager or by the distributor which you deal with. Configuration is so simple as the below link
http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a00803f82eb.shtml
2-If there is no budget , use B-ACD is a TCL and you can download required files from Cisco site if these files not on your flash. Please find the below link for B-ACD configuration
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40tclov.html
Thanks
Please rate all useful information -
CUCM 7 - Route incoming call to a specific VM
Hello Forum,
Hope this is an easy one for you guys.
I am trying to accomplish the following and not sure if its possible.
Ex.
Incoming call to ext. 1111
Ext. 1111 gets added to 2 phones, then call rings simultaneously on said 2 phones
After 5 rings and no answer, it rolls over to a different already existing VM (ext 8888)
Any help would be appreciated as I am not sure where to change the settings to make this happen.
Thanks in advance,
LSAssuming x1111 does not need a dedicated voicemail, you could do the following:
Ensure that x1111 has the correct VM profile and set the CFB/CFNA settings to forward to VM as usual. On the mailbox for 8888, set an alternate extension of 1111.
Hailey
Please rate helpful posts! -
PAP2T: Incoming calls being blocked by Router.
Hi,
I am trying to use an unlocked PAP2T Adapter with Telus home network in Canada and am not able to receive calls. I can make outgoing calls to any number in the world, but whenever I get a call back, the phone doesn't ring & later I find it in Firewall log as being blocked under "Low Risk Attack". While trying to find a resolution with the VOIP provider I had at times picked up the phone randomly & found the caller on the line. So apparently the calls have no prblem reaching my PAP2T Adapater, it is just that (1) there is no ring when they do & (2) the incoming calls get blocked by Firewall & dropped.
I have tried port forwarding (UDP & TCP) as well DMZ mode for the Adapter, no luck. Later i tried disabling almost all of the internet provider's 2wire modem/router firewall features (UDP ports scan, packets , etc.) still didnt work. I have run out of ideas & hoping that someone can help me find a way out of this problem ? Appreciate your help & time. Thanks.I changed to the following settings that did the trick for making the phone ring,
Click Regional on the top:
Under Ring and Call Waiting Tone Spec
Change Ring Waveform to Sinusoid
Change Ring Voltage: 90
Change Ring Frequency: 20
However, when I pick up the ringing phone, there is no voice going through the line. Also, as per my VOIP provider instructions, when I dial a local number followed by # sing & hang up in 5 seconds, I don't get any callback either. I have even updated the PAP2T to the latest firmware 5.1.6. Any thing else I can change to get the voice across & get the call back ? thanks for your help again. -
Sir recd your reply my router is perfect at the same time my I phone is receving incoming call
Is there a question here?
-
Iphone 4S; Phone & audio piece; Incoming calls are NOT routed automatically??
When I dial out; no problem the calls automatically goes through ear pice.
When I get incomming calls the call goes to phone, not even to the speaker in in the phone. So I have to
1.Slide the bar (then first realizing (again) dam: I hear nothing)
2. Choose the box for speaker options
3. Choose my blue ant piece..
By then some callers think I am not picking up or there is somthing wrong..!
My 3 colleaguers have same phone and no such problems. We are at a loss and I could not find reply in community groups. There was note about Iphone 5 and not correcting to Car speaker automatically. This was noted as a softwre issue Apple might address in next version. But since my colleagues do not have problem with their Iphone it must be either my settings or a weird fault with my phone.
One of my collegues have same ear pice as me, so the issueis not that either.
Is there some weird setting I need to fix to correct the problem?
Thank youAre you able to make calls? I would try a couple different things -
1. Try doing a double hard reset - Press and hold home button and the sleep wake button at the same time until the screen displays the slide to shut down red button. Continue pressing and holding both the buttons until the screen goes black and then the apple logo flashes up. Continue pressing and holding both the home and sleep wake button beyond this point until the screen goes black again. At this point, let go of the home button and press once the sleep wake button as you would to start your phone. See if this double hard reset helps clear out any cache.
2. If this fails, I would try restoring the phone as a new phone in iTunes. Backup your phone before this so that you dont lose your contacts, etc. You can do a couple test calls once the firmware is installed before setting up the phone, data, apps, etc. If it looks good, try restoring your phone from an earlier backup.
If everything fails, call AppleCare or see a Genius in the store. -
Switched over to MGCP from H323, no incoming calls fast busy
Hello, I'm on the network side crossing over to the Voice side. We replaced a 3825 Voice Router at a branch office with a 2921. The 3825 was setup with a T1 and had a PRI connected to the FXO ports. The 2921 is now connected via fiber and TAC helped get the router registered to the Call Manager. I'm trying to match up the old dial peers on the new router. I can't make out going or receive incoming calls, I get a This Call Can't be completed at this time.
When it was on the T1, the branch office was using H323. Now that's connects to the same CUCM, it's on MGCP. Shouldn't the old Dial Peers work on the new router?I had to configure the FXO ports for the DN to route the incoming calls. I learned just because the Call Manager configured the Voice Gateway as a client, you still must configure the FXO ports to route the main DID to a DN on the LAN.
Great advice, learning alot about telephony and VoIP.
Wish I had more experience troubleshooting the CUCM and DID portion, also learned the phone compnay doesn't turn up their ISDN switch until you configure your PBX. And a PRI testing isn't the same as the data T1, it's about having the Signaling channel turned up and or configured.
Lessons learned, wished I would have crossed over to VoIP earlier. -
Help! Why is the Pre's speaker ringing an incoming call when I have a headset on?
Hi all,
I just had the most embarrassing thing happen to me today.
I was listening to some music on my Pre with the headset (that came with the Pre), and I received an incoming call.
I heard it ring and answered the call, no problem what I thought!
When I got off the phone, my co-worker complained that I had the volume on the Pre too loud. I looked at her and said WTF, I said you heard the phone ring. She said yes!
I then did a test, I exited the music playing, and opened the phone app and make a phone call out. So I made sure that the icon in the phone app showed that I was using the headset.
Then I ended that call, and used a land line and call myself on the Pre.
The Pre rang both in the headset and also the speaker the Pre?
Why is the Pre using the speaker when the phone rings if I'm using a headset?
Isn't the purpose of the headset to provide privacy and keep my neighbors from being disturbed?
I read about the switch turning off the ring but not the music, but this wasn't a case where I needed to use the switch, I had the headphones on.
Why is this happening, is this a bug ... or is there some sort of setting that I don't know about?
Thanks!
Rob
Post relates to: Pre p100eww (Sprint)I'm having the same problem. I had a plantronics 975 (which I lost, sadly). When connected, the ringer would always be routed through the bluetooth and the iPhone's external speaker. When in silent mode, the iPhone would only vibrate, but I could still hear the ringer in the bluetooth.
I just bought a new Plantronics Voyager+. Now when it's connected, all sounds are routed to the earpiece - including the taps, etc. The iPhone's speaker doesn't ring at all, unless the bluetooth is disconnected or powered off. I'd really like to be able to hear both, like I was before. This is an iPhone 3G running iOS 4 (latest version, not sure what it is exactly). Haven't updated software/firmware in between bluetooths, just changed the earpiece. -
CME\7960 running SIP firmware - How do i setup incoming calls? - Can anyone help please?
Hi Guys,
I have a SIP trunk setup with a 2811 running CME version 7. I can make outbound calls ok but having issues getting the incoming calls working, i have 1 number on my SIP trunk and that is 01133501788 and i want that to ring my Cisco 7960 which is running SIP firmware not SCCP. I have included by config for anyone who can help me, i just want the incoming call to work.
Many Thanks.
Matthew.
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone GMT 0
dot11 syslog
ip source-route
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.1.1
ip dhcp excluded-address 10.10.10.1
ip dhcp pool DATA_POOL
network 10.10.10.0 255.255.255.0
default-router 10.10.10.1
dns-server 188.92.232.50 188.92.232.100
ip dhcp pool VOICE_POOL
network 192.168.1.0 255.255.255.0
default-router 192.168.1.1
dns-server 188.92.232.50 188.92.232.100
option 150 ip 192.168.1.1
ip name-server 188.92.232.50
ip name-server 188.92.232.100
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
bind control source-interface FastEthernet0/1.20
bind media source-interface FastEthernet0/1.20
registrar server
voice class codec 1
codec preference 2 g711ulaw
codec preference 3 g711alaw
voice register global
mode cme
source-address 192.168.1.1 port 5060
max-dn 144
max-pool 42
load 7960-7940 P0S3-8-12-00
authenticate register
tftp-path flash:
create profile sync 0008072514198272
voice register dn 1
number 6999
allow watch
name SIP
label SIP
voice register pool 1
id mac 000F.902B.40E0
type 7960
number 1 dn 1
dtmf-relay sip-notify
username cisco password cisco
codec g711ulaw
voice translation-rule 1
rule 1 /^9\(.*\)/ /\1/
voice translation-rule 2
rule 1 /^6...$/ /4143*002/
voice translation-profile DiscardDigit9
translate calling 2
translate called 1
voice translation-profile IncomingSIP
translate calling 1133501788
voice-card 0
no dspfarm
username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
archive
log config
hidekeys
interface FastEthernet0/0
ip address 194.12.0.222 255.255.255.252
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1
no ip address
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1.10
description DATA
encapsulation dot1Q 10
ip address 10.10.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/1.20
description VOICE
encapsulation dot1Q 20
ip address 192.168.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 194.12.0.221
ip http server
ip http authentication local
no ip http secure-server
ip nat inside source list 1 interface FastEthernet0/0 overload
access-list 1 permit 192.168.1.0 0.0.0.255
access-list 1 permit 10.10.10.0 0.0.0.255
tftp-server flash:P003-8-12-00.bin
tftp-server flash:P003-8-12-00.sbn
tftp-server flash:P0S3-8-12-00.loads
tftp-server flash:P0S3-8-12-00.sb2
tftp-server flash:P003-8-12-00
tftp-server flash:P003-8-12-00.loads
tftp-server flash:P003-8-12-00.sb2
tftp-server flash:SIP000F902B40E0.cnf.xml
control-plane
mgcp behavior g729-variants static-pt
dial-peer cor custom
dial-peer voice 2 voip
description Outgoing Geographic
translation-profile outgoing DiscardDigit9
destination-pattern 0[7]........
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
dtmf-relay rtp-nte
no vad
dial-peer voice 1 voip
description IncomingSIP
translation-profile incoming IncomingSIP
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
sip-ua
credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
authentication username 4143*002 password 7 password
nat symmetric role passive
nat symmetric check-media-src
calling-info sip-to-pstn number set 4143*002
no remote-party-id
retry invite 3
retry register 3
timers connect 100
registrar dns:sip.cloudcalling.co.uk expires 60
sip-server dns:sip.cloudcalling.co.uk
host-registrar
gatekeeper
shutdown
telephony-service
load 7960-7940 P0S3-8-12-00
max-ephones 24
max-dn 30
ip source-address 192.168.1.1 port 2000
max-conferences 8 gain -6
web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
line con 0
line aux 0
line vty 0 4
login
scheduler allocate 20000 1000
ntp server 85.119.80.232
end
Router#You my friend are a star! worked straight away, many thanks. Just one more thing, when i make an outgoing call, it always appears as "blocked" on my phone, my sip trunk is set to allow CME to alter outgoing CLI's how would i program the outgoing CLI to 01133501788 also?
The new working config is below with your suggestion, which works!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone GMT 0
dot11 syslog
ip source-route
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.1.1
ip dhcp excluded-address 10.10.10.1
ip dhcp pool DATA_POOL
network 10.10.10.0 255.255.255.0
default-router 10.10.10.1
dns-server 188.92.232.50 188.92.232.100
ip dhcp pool VOICE_POOL
network 192.168.1.0 255.255.255.0
default-router 192.168.1.1
dns-server 188.92.232.50 188.92.232.100
option 150 ip 192.168.1.1
ip name-server 188.92.232.50
ip name-server 188.92.232.100
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server
voice class codec 1
codec preference 2 g711ulaw
codec preference 3 g711alaw
voice register global
mode cme
source-address 192.168.1.1 port 5060
max-dn 144
max-pool 42
load 7960-7940 P0S3-8-12-00
authenticate register
tftp-path flash:
create profile sync 0015244443466064
voice register dn 1
number 6999
allow watch
name SIP
label SIP
voice register pool 1
id mac 000F.902B.40E0
type 7960
number 1 dn 1
dtmf-relay sip-notify
username cisco password cisco
codec g711ulaw
voice translation-rule 1
rule 1 /^6...$/ /4143*002/
voice translation-rule 3
rule 1 /^01133501788$/ /6999/
rule 2 /^1133501788$/ /6999/
voice translation-profile IncomingSIP
translate called 3
voice translation-profile Translatetrunk
translate calling 1
voice-card 0
no dspfarm
username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
archive
log config
hidekeys
interface FastEthernet0/0
ip address 194.12.0.222 255.255.255.252
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1
no ip address
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1.10
description DATA
encapsulation dot1Q 10
ip address 10.10.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/1.20
description VOICE
encapsulation dot1Q 20
ip address 192.168.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 194.12.0.221
ip http server
ip http authentication local
no ip http secure-server
ip nat inside source list 1 interface FastEthernet0/0 overload
access-list 1 permit 192.168.1.0 0.0.0.255
access-list 1 permit 10.10.10.0 0.0.0.255
tftp-server flash:P003-8-12-00.bin
tftp-server flash:P003-8-12-00.sbn
tftp-server flash:P0S3-8-12-00.loads
tftp-server flash:P0S3-8-12-00.sb2
tftp-server flash:P003-8-12-00
tftp-server flash:P003-8-12-00.loads
tftp-server flash:P003-8-12-00.sb2
tftp-server flash:SIP000F902B40E0.cnf.xml
control-plane
mgcp behavior g729-variants static-pt
dial-peer cor custom
dial-peer voice 1 voip
description IncomingSIP
translation-profile incoming IncomingSIP
voice-class codec 1
session protocol sipv2
session target sip-server
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
dial-peer voice 2 voip
description Outgoing Geographic
translation-profile outgoing Translatetrunk
destination-pattern 0[7]........
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
dtmf-relay rtp-nte
no vad
sip-ua
credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
authentication username 4143*002 password 7 password
nat symmetric role passive
nat symmetric check-media-src
calling-info sip-to-pstn number set 4143*002
no remote-party-id
retry invite 3
retry register 3
timers connect 100
registrar dns:sip.cloudcalling.co.uk expires 60
sip-server dns:sip.cloudcalling.co.uk
host-registrar
gatekeeper
shutdown
telephony-service
load 7960-7940 P0S3-8-12-00
max-ephones 24
max-dn 30
ip source-address 192.168.1.1 port 2000
max-conferences 8 gain -6
web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
transfer-system full-consult
create cnf-files version-stamp 7960 Dec 17 2013 14:35:13
line con 0
line aux 0
line vty 0 4
login
scheduler allocate 20000 1000
ntp server 85.119.80.232
end
Router# -
SIP incoming call, won't work (CME)
Hi all,
I'm facing a weird problem and the sip-provider can't help. I suppose there is a problem with the dial-peer/translation-rule but I can't figure it out...
There is a CME (c2800nm-ipvoice-mz.124-11.XW10.bin, CME Version 4.2(0)) with a
SIP trunk.
Outgoing calls are working (DID).
Incoming calls (all DID) are ringing on the same internal number.
The situation:
- external call on 0815440097 is ringing on the internal nr. 296 (should be 297)
- external call on 0815440096 is ringing on the internal nr. 296
Here the config:
================================
voice service voip
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
no update-callerid
voice translation-rule 40
rule 2 /\(.*\)/ /9\1/
voice translation-rule 190
rule 1 /^0\(.*\)/ /\1/
rule 2 /^9\(.*\)/ /\1/
voice translation-rule 191
rule 2 /296/ /0815440096/
rule 3 /297/ /0815440097/
voice translation-rule 192
rule 2 /^0815440097/ /297/
rule 3 /^0815440096/ /296/
voice translation-profile TP_IN_SIP
translate calling 40
translate called 192
voice translation-profile TP_OUT_SIP
translate calling 191
translate called 190
dial-peer voice 2000 voip
description *** SIP-TRUNK (IN/OUT) ***
translation-profile incoming TP_IN_SIP
translation-profile outgoing TP_OUT_SIP
destination-pattern 0.T
b2bua
session protocol sipv2
session target dns:sip12.e-fon.ch
session transport udp
incoming called-number 0815440096
dtmf-relay rtp-nte
codec g711alaw
no vad
sip-ua
credentials username 0815440096 password 7 xxxx realm sip12.e-fon.ch
keepalive target dns:sip12.e-fon.ch
authentication username 0815440096 password 7 xxxx
calling-info pstn-to-sip from number set 0815440096
no remote-party-id
retry invite 2
retry response 2
retry bye 2
retry register 2
retry options 1
registrar dns:sip12.e-fon.ch expires 69
sip-server dns:sip12.e-fon.ch
reason-header override
connection-reuse
host-registrar
sh sip-ua register status
Line peer expires(sec) registered
================================ ========== ============ ==========
0815440096 20005 18 yes
Here the CCSIP MESSAGE debug (looks ok):
(call from 0000000000 to 0815440097)
===============================
Mar 8 21:55:10.469 METD: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:212.55.198.132;lr=on;ftag=as00cd0e7f>
Via: SIP/2.0/UDP 212.55.198.132;branch=z9hG4bK49ff.2d35e30a71291ffe3895b39164900f36.0
Via: SIP/2.0/UDP 212.55.198.134:5061;branch=z9hG4bK1cb84749;rport=5061
Max-Forwards: 69
From: "0000000000" <sip:[email protected]:5061>;tag=as00cd0e7f
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: e-fon
Date: Thu, 08 Mar 2012 20:55:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-IPCONNECT: 0815440097
X-Number: 0815440097
Content-Type: application/sdp
Content-Length: 415
v=0
o=root 770254981 770254981 IN IP4 212.55.198.134
s=Asterisk PBX 1.6.1.20
c=IN IP4 212.55.198.134
t=0 0
m=audio 11886 RTP/AVP 8 9 111 3 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Mar 8 21:55:10.481 METD: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.55.198.132;branch=z9hG4bK49ff.2d35e30a71291ffe3895b39164900f36.0,
Via:SIP/2.0/UDP 212.55.198.134:5061;branch=z9hG4bK1cb84749;rport=5061
From: "0000000000" <sip:[email protected]:5061>;tag=as00cd0e7f
To: <sip:[email protected]:5060>
Date: Thu, 08 Mar 2012 20:55:10 GMT
Call-ID: [email protected]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
Here is the VOICE DIAL-PEER debug (call from 0000000000 to 0815440097):
=============================================
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=0815440096, Called Number=0815440096, Peer Info
Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=0815440096
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20005
2: Dial-peer Tag=2000
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=0000000000, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2000
Mar 8 22:00:09.502 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=0000000000, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.502 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2000
Mar 8 22:00:09.502 METD: //-1/8647979A82E1/DPM/dpAssociateIncomingPeerCore:
Calling Number=0000000000, Called Number=0815440096, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.502 METD: //-1/8647979A82E1/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2000
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=0815440096, Called Number=0815440096, Peer Info
Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=0815440096
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20006
2: Dial-peer Tag=2000
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=0815440096, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ANSWER; Incoming Dial-peer=2000
Mar 8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersCore:
Calling Number=, Called Number=0815440096, Peer Info
Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=0815440096
Mar 8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Mar 8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20006
2: Dial-peer Tag=2000
show dial-peer voice summary:
dial-peer hunt 0
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT
PORT
555 voip up up 555 0 syst loopback:rtp
20001 pots up up 296$ 0 50/0/1
20002 pots up up 297$ 0 50/0/2
2000 voip up up 0.T 0 syst dns:sip12.e-fon.ch
20005 pots up up 0815440096$ 0 50/0/150
20006 pots up up 0815440097$ 9 50/0/2
voip translation debugging (call from 0794142975 to 0815440097):
=========================================
Mar 8 22:35:26.145 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFCA0; count=1
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFCA0; count=0
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: number=0794142975 type=unknown plan=unknown numbertype=calling
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/sed_subst: Successful substitution; pattern=0794142975 matchPattern=(.*) replacePattern=9\1 replaced pattern=90794142975
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: xlt_number=90794142975 xlt_type=unknown xlt_plan=unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-called
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-called number not found
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: number=0815440096 type=unknown plan=unknown numbertype=called
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=3
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: No match found
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: No match: number=0815440096 type=unknown plan=unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFCA0; count=1
Mar 8 22:35:26.153 METD: //-1/73E51DB2834F/RXRULE/regxrule_dp_translate: No profile found in peer 20005 for outgoing direction
Mar 8 22:35:26.153 METD: //-1/73E51DB2834F/RXRULE/regxrule_dp_translate: calling_number=90794142975 calling_octet=0x0
called_number=0815440096 called_octet=0x0
redirect_number= redirect_type=0 redirect_plan=0 redirect_PI=-1 redirect_SI=-1
Mar 8 22:35:26.181 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFCA0; count=2
Thanks,
NorbertHi Alex,
Thank you for the reply.
After changing the "incoming called-number" I got the same output.
The weird think is, why the dial-peer debug shows the 0815440096 number, despite the right "to: number" in the SIP-Message.
Is there a problem with the "voice service voip" or "sip-ua"?
on the voice translation debug I see:
Match Rule=DP_MATCH_TO_URI; URI=sip:0815440097
Match Rule=DP_MATCH_FROM_URI; URI=sip:0819262424
But I guess the translation rule is maching this one:
Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=0815440096
So how can the voice translation rule be set to map the entry DP_MATCH_TO_URI; URI=sip:0815440097
Thanks for the help.
Regards,
Norbert
voip translation debugging (call from 0819262424 to 0815440097):
===================================================
Mar 9 07:45:16.371 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFAFC; count=1
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFAFC; count=0
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: number=0819262424 type=unknown plan=unknown numbertype=calling
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/sed_subst: Successful substitution; pattern=0819262424 matchPattern=(.*) replacePattern=9\1 replaced pattern=90819262424
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: xlt_number=90819262424 xlt_type=unknown xlt_plan=unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-called
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-called number not found
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: number=0815440096 type=unknown plan=unknown numbertype=called
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 3 in ruleset 192
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 3 in ruleset 192
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/sed_subst: Successful substitution; pattern=0815440096 matchPattern=^0815440096 replacePattern=296 replaced pattern=296
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: xlt_number=296 xlt_type=unknown xlt_plan=unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFAFC; count=1
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_dp_translate: No profile found in peer 20001 for outgoing direction
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_dp_translate: calling_number=90819262424 calling_octet=0x0
called_number=296 called_octet=0x0
redirect_number= redirect_type=0 redirect_plan=0 redirect_PI=-1 redirect_SI=-1
Mar 9 07:45:16.379 METD: //-1/439ABF97847F/RXRULE/regxrule_vp_translate: No profile found in voice port or trunk group for outgoing direction
Mar 9 07:45:16.379 METD: //-1/439ABF97847F/RXRULE/regxrule_vp_translate: calling_number=90819262424 calling_octet=0x0
called_number=296 called_octet=0x0
redirect_number= redirect_type=0 redirect_plan=0
Mar 9 07:45:18.195 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFAFC; count=2
debug voice dialpeer detail
=====================
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=0815440096, Expanded String=0815440096, Calling Number=0815440096T
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=2000 Is Matched
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=20005 Is Matched
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=0819262424
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=0819262424T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ORIGINATE; Calling Number=0819262424
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=0819262424T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=2000 Is Matched
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=0819262424
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=0819262424T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ORIGINATE; Calling Number=0819262424
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=0819262424T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=2000 Is Matched
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_REQUEST_URI; URI=sip:[email protected]:5060
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_TO_URI; URI=sip:[email protected]:5060
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_FROM_URI; URI=sip:[email protected]:5061
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=0815440096
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=0815440096, Expanded String=0815440096, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=0819262424
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=0819262424T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ORIGINATE; Calling Number=0819262424
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=0819262424T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=2000 Is Matched
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=296, Expanded String=296, Calling Number=296T
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=20001 Is Matched
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=296
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=296T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ORIGINATE; Calling Number=296
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=296T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=20001 Is Matched
Mar 9 07:49:25.584 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=296, Expanded String=296, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.584 METD: //-1/D8245087848D/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=20001 Is Matched
Mar 9 07:49:25.588 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=90819262424, Expanded String=90819262424, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.588 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=0819262424, Expanded String=0819262424, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=2000 Is Matched
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=296
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=296T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ORIGINATE; Calling Number=296
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=296T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=20001 Is Matched
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=296, Expanded String=296, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=20001 Is Matched
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=90819262424, Expanded String=90819262424, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=90819262424, Expanded String=90819262424, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=296, Expanded String=296, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=20001 Is Matched -
Matching B-channel for dialpeer assigment (Incoming call)
Is it possible to manipulate a specific B Channel to use a dialpeer? For example on a T1 with 24 channels, I need the last 4 B-channels to choose an specific dialpeer, same concepts as matching incoming DID's but on a B-Channel slot.
If a call arrives on B-channel 20, I need that call to be sent to a specific DN at the CallManager.
I am using H.323 on the gateway side.
thanks in advanced.
OscarHi!
I'm stuck in a similar situation at the moment, and hope someone has solved this issue.
I have a "back to back connection" between two PBX'es that communicate using Q.931. I have had to replace my old hardware running a legacy CCS solution due to its incapability of understanding overlap signalling. I have replaced it with two Cisco routers running E1 (Q931). Between the units I have a high latency low bandwidth network.
The issue is that the PBX'es are configured in a way that requires calls between the PBX'es to use the same time slot at both ends. I'm running a VoIP network with CUCM between the two sites.
=============================================
controller E1 0/2/0
framing NO-CRC4
pri-group timeslots 1-21
trunk-group TS01 timeslots 1
trunk-group TS02 timeslots 2
trunk-group TS03 timeslots 3
trunk-group TS04 timeslots 4
trunk-group TS05 timeslots 5
trunk-group TS06 timeslots 6
trunk-group TS07 timeslots 7
trunk-group TS08 timeslots 8
trunk-group TS09 timeslots 9
trunk-group TS10 timeslots 10
trunk-group TS11 timeslots 11
trunk-group TS12 timeslots 12
trunk-group TS13 timeslots 13
trunk-group TS14 timeslots 14
trunk-group TS15 timeslots 15
trunk-group TS17 timeslots 17
trunk-group TS18 timeslots 18
trunk-group TS19 timeslots 19
trunk-group TS20 timeslots 20
trunk-group TS21 timeslots 21
dial-peer voice 81030001 pots
trunkgroup TS01
description ** PBX TS01 **
translation-profile incoming 61031401
translation-profile outgoing 25
destination-pattern 81030001T
progress_ind alert enable 8
progress_ind progress enable 2
incoming called-number .
no digit-strip
dial-peer voice 81030002 pots
trunkgroup TS02
description ** PBX TS02 **
translation-profile incoming 61031402
translation-profile outgoing 25
destination-pattern 81030002T
progress_ind alert enable 8
progress_ind progress enable 2
incoming called-number .
no digit-strip
voice translation-rule 25
rule 1 /^810300../ //
voice translation-rule 61031401
rule 1 // /61031401\1/
voice translation-rule 61031402
rule 1 // /61031402\1/
voice translation-profile 25
translate called 25
voice translation-profile 61031401
translate called 61031401
voice translation-profile 61031402
translate called 61031402
=====================================
The idea is that I do not know (or care) what numbers are used as SOURCE or DESTINATION of the original call. My network should be transparent to the PBX number plan. I need to add a prefix, and it should be based on the timeslot the call comes in on. I route the traffic between the routers using the prefix.
The configuration excerpt above should add 61031401 prefix to all calls entering on TS01, and 61031402 to all calls entering on TS02 etc. Calls from the remote should have corresponding prefixes 81030001 for TS01 and 81030002 for TS02 etc.
The outbound (from voip to pots) routing of the above configuration works.
However I have a challenge with the incoming prefixing.
All calls inbound end up using "dial-peer 81030001 pots".
I believe the reason this dial-peer "takes" all of the calls inbound from pots is due to the line "incoming called-number ."
Removing this makes no inbound pots call work as the "destination-pattern 8103001T" is never matched.
Removing "destination-pattern 8103001T" from the dial-peer is not working as it kills the voip to pots routing of inbound calls from the remote router.
Anyone got a good idea for me? -
000049: Oct 26 14:51:04.181: %VOICE_IEC-3-GW: H323: Internal Error (H323 Interwo
rking Error): IEC=1.1.127.5.21.0 on callID 261 GUID=80316352D0FE011E02002302A5CC
B8F9
000050: Oct 26 15:24:50.315: %CALL_CONTROL-6-CALL_LOOP: The incoming call has a
global identfier already present in the list of currently handled calls. It is b
eing refused.
000051: Oct 26 15:24:50.315: %VOICE_IEC-3-GW: CCAPI: Internal Error (Incoming lo
op): IEC=1.1.180.1.28.0 on callID 0
000052: Oct 26 15:24:50.315: %VOICE_IEC-3-GW: H323: Internal Error (H323 Interwo
rking Error): IEC=1.1.127.5.21.0 on callID 363
According to Cisco it says:
It means that the voice gateway has detected a loop in the call route
What debugs can I do to track this problem down.
All dial peers are like this. They are all voip dial peers with different destination-patterns and different session target IP addresses. No PSTN dial peers at all.
dial-peer voice 30000 voip
huntstop
destination-pattern 3[01]...
session target ipv4:192.168.99.4
dtmf-relay h245-signal
ip qos dscp cs5 mediaQ: How long has this system been in place - is it a new system, or is this something that just started on a system that has been in place for awhile?
A: My customer says to me that this system has been in place for nearly a year and the problem started only a couple of days ago.
Q: If so were there any changes to the infrastructure, or the dialplan in particular?
A: Again this is the first question I asked and he said “NO CHANGES”. But that is what most customer say – right?
Q: Have you seen this occur in the past in the logs? If so how often do you see it?
A: Started on 26th Oct 2010. Since the problem started the router has been reloaded and the problem is still there. It is very intermittent and I have to do the debug when I get the problem.
Q: Have you tried to establish what is causing these: H323: Internal Error (H323 Interworking Error) ?
A: Not sure how to troubleshoot this. The explanation of this message on CCO is not that great. -
Incoming called URI number manipulation in Call manager 10.5
Dear Experts,
can we manipulate the called URI number like we manipulate the digits (e.g Translation pattern) ?
can we have manipulate the incoming called number to match a route pattern
for eg. the called uri is 955XXXX@CUCM-address
route pattern 955XXXX ==> Voice Gateway
thanks for your help in advance
Anasbecause it should match a route pattern not Directory ? it always return 404 not found
it comes as a URI because it is use SIP trunk to reach the CUCM -
Analog line (FXO) Incoming calls getting connected after 3 rings
HI,
we are having 4 Analog line (FXO)...Every time when callers call the number they hear 3 rings & after that call frwds to AA or any extension.
In show voice port summary, we can see that voice port is getting connect at the first ring but after 3 rings only phone rings.
here is the o/p of voice port.
Foreign Exchange Office 0/0/0 Slot is 0, Sub-unit is 0, Port is 0
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 3 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 128 ms
Echo Cancel worst case ERL is set to 6 dB
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is plar
Connection Number is 250
Initial Time Out is set to 15 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Power Denial Disconnect Time Out is set to 1000 ms
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 30 s
Companding Type is u-law
Region Tone is set for AE
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Station name None, Station number None
Caller ID Info Follows:
Standard BELLCORE
Caller ID is received after 1 ring(s)
Translation profile (Incoming): INCOMING_CallerID_PROFILE
Translation profile (Outgoing):
lpcor (Incoming):
lpcor (Outgoing):
Voice card specific Info Follows:
Signal Type is loopStart
Battery-Reversal is enabled
Number Of Rings is set to 1
Supervisory Disconnect is signal
Answer Supervision is inactive
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Out Type is dtmf
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Percent Break of Pulse is 65 percent
GuardOut timer is 2000 ms
Minimum ring duration timer is 125 ms
Hookflash-in Timing is set to 600 ms
Hookflash-out Timing is set to 400 ms
Supervisory Disconnect Timing (loopStart only) is set to 350 ms
OPX Ring Wait Timing is set to 6000 ms
Secondary dialtone is disabledhostname VGUAE001
no aaa new-model
clock timezone UAE 4 0
ip cef
ip domain name yourdomain.com
no ipv6 cef
multilink bundle-name authenticated
trunk group ALL_FXO
max-retry 5
voice-class cause-code 1
hunt-scheme longest-idle
translation-profile outgoing PROFILE_ALL_FXO
voice-card 0
voice call send-alert
voice rtp send-recv
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
voice class cause-code 1
no-circuit
voice translation-rule 1112
rule 1 /^9/ //
voice translation-rule 3265
rule 1 // /9\1/
voice translation-profile INCOMING_CallerID_PROFILE
translate calling 50
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
license udi pid CISCO2901/K9 sn FCZ173992Z8
hw-module pvdm 0/0
hw-module pvdm 0/1
username cisco privilege 15 secret 4 opjnnkXqCr4kCOa9DuALcNpBOMetBAc/usnpSWADsCI
username godiva privilege 15 secret 4 cH8b8z.ioYu/pMv/AKuEcBd/f6g9v/vm/s3aXeqUAd6
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.31.2 255.255.255.0
ip helper-address 192.168.31.11
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.31.2
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
ip route 0.0.0.0 0.0.0.0 192.168.31.1
control-plane
voice-port 0/0/0
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
groundstart auto-tip
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/1
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/2
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/3
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 250
caller-id enable
mgcp profile default
dial-peer voice 2000 voip
destination-pattern 2..
session target ipv4:192.168.31.11
incoming called-number .
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 10 pots
trunkgroup ALL_FXO
description **CCA*UAE*Fire**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 997
forward-digits all
no sip-register
dial-peer voice 11 pots
trunkgroup ALL_FXO
description **CCA*UAE*International Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 900T
forward-digits all
no sip-register
dial-peer voice 12 pots
trunkgroup ALL_FXO
description **CCA*UAE*Eitisalat**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9101
forward-digits all
no sip-register
dial-peer voice 13 pots
trunkgroup ALL_FXO
description **CCA*UAE*Water or electrical emergencies**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 971
forward-digits all
no sip-register
dial-peer voice 14 pots
trunkgroup ALL_FXO
description **CCA*UAE*Police and emergencies**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 999
forward-digits all
no sip-register
dial-peer voice 15 pots
trunkgroup ALL_FXO
description **CCA*UAE*National area codes**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[1-579].......
forward-digits all
no sip-register
dial-peer voice 16 pots
trunkgroup ALL_FXO
description **CCA*UAE*Mobile Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 90[5-6][0-7].......
forward-digits all
no sip-register
dial-peer voice 17 pots
trunkgroup ALL_FXO
description **CCA*UAE*toll-free**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[2-9]00T
forward-digits all
no sip-register
dial-peer voice 18 pots
trunkgroup ALL_FXO
description **CCA*UAE*Fixed Line Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[2-8]T
forward-digits all
no sip-register
dial-peer voice 19 pots
trunkgroup ALL_FXO
description **CCA*UAE*808**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9808T
forward-digits all
no sip-register
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/1
dial-peer voice 52 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/2
dial-peer voice 53 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/3
dial-peer voice 54 pots
description ** FXO pots dial-peer **
destination-pattern A0
port 0/0/0
no sip-register
dial-peer voice 55 pots
description ** FXO pots dial-peer **
destination-pattern A1
port 0/0/1
no sip-register
dial-peer voice 56 pots
description ** FXO pots dial-peer **
destination-pattern A2
port 0/0/2
no sip-register
dial-peer voice 57 pots
description ** FXO pots dial-peer **
destination-pattern A3
port 0/0/3
no sip-register
Debug vpm signal:
Nov 23 19:31:31.556: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing
Nov 23 19:31:31.556: htsp_timer - 125 msec
Nov 23 19:31:31.684: htsp_process_event: [0/0/0, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer
Nov 23 19:31:31.684: htsp_timer - 10000 msec
Nov 23 19:31:31.684: htsp_timer3 - 5600 msec
Nov 23 19:31:31.684: [0/0/0] htsp_start_caller_id_rx:Mode BELLCORE. Alerting 0x1
Nov 23 19:31:31.684: htsp_start_caller_id_rx create dsp_stream_manager
Nov 23 19:31:31.684: [0/0/0] htsp_dsm_create_success returns 1
Nov 23 19:31:33.604: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
Nov 23 19:31:33.604: fxols_ringing_not
Nov 23 19:31:33.604: htsp_timer_stop
Nov 23 19:31:33.604: htsp_timer - 10000 msec
Nov 23 19:31:37.284: htsp_process_event: [0/0/0, FXOLS_RINGING, E_HTSP_EVENT_TIMER3]fxols_snoop_clid_stop
Nov 23 19:31:37.284: htsp_timer_stop3
Nov 23 19:31:37.516: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0000]
Nov 23 19:31:39.604: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
Nov 23 19:31:39.604: fxols_ringing_not
Nov 23 19:31:39.604: htsp_timer_stop
Nov 23 19:31:39.604: htsp_timer_stop3
Nov 23 19:31:39.604: [0/0/0] htsp_stop_caller_id_rx. message length 0htsp_setup_ind
Nov 23 19:31:39.604: [0/0/0] get_fxo_caller_id:Caller ID receive failed. parseCallerIDString:no data.
Nov 23 19:31:39.604: [0/0/0] get_local_station_id calling num= calling name= calling time=11/23 23:31 orig called=
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=250
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/cc_api_call_setup_ind_common:
Interface=0x3CE27724, Call Info(
Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=250(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE,
Incoming Dial-peer=50, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=FALSE,
Source Trkgrp Route Label=ALL_FXO, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/ccCheckClipClir:
In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/ccCheckClipClir:
Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: :cc_get_feature_vsa malloc success
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: cc_get_feature_vsa count is 1
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: :FEATURE_VSA attributes are: feature_name:0,feature_time:1025218944,feature_id:83
Nov 23 19:31:39.604: //83/B583C95F8093/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=250(TON=Unknown, NPI=Unknown))
Nov 23 19:31:39.608: [0/0/0] htsp_dsm_close_done
Nov 23 19:31:39.608: htsp_process_event: [0/0/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
Nov 23 19:31:39.608: fxols_wait_setup_ack:
Nov 23 19:31:39.608: [0/0/0] set signal state = 0xC timestamp = 0fxols_check_auto_call
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/cc_process_call_setup_ind:
Event=0x22ACD828
Nov 23 19:31:39.608: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 250
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetContext:
Context=0x230F9C10
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 83 with tag 50 to app "_ManagedAppProcess_Default"
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=FALSE, Mode=0,
Outgoing Dial-peer=2000, Params=0x230FB0D0, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCheckClipClir:
In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCheckClipClir:
Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
Destination Pattern=2.., Called Number=250, Digit Strip=FALSE
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=250(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=
Account Number=, Final Destination Flag=TRUE,
Guid=B583C95F-53AC-11E3-8093-C8EEBDE4256A, Outgoing Dial-peer=2000
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=250
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x22847B14, Interface Type=1, Destination=, Mode=0x0,
Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=250(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE, Outgoing Dial-peer=2000, Call Count On=FALSE,
Source Trkgrp Route Label=ALL_FXO, Target Trkgrp Route Label=, tg_label_flag=1, Application Call Id=)
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: :cc_get_feature_vsa malloc success
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: cc_get_feature_vsa count is 2
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: :FEATURE_VSA attributes are: feature_name:0,feature_time:1025218720,feature_id:84
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=1, FlowMode=1
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccCallSetContext:
Context=0x230FB080
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=2000
Nov 23 19:31:39.612: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_proc
Nov 23 19:31:39.612: htsp_timer - 120000 msec
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccGetMediaClassTag:
media class tag 0
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccSetMediaclassIp2ipTags:
media class tags set: NR 0, ASP 0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccSetMediaclassIp2ipTags:
media class tags set: NR 0, ASP 0
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccGet_xc_nr_asp_info:
media class tags: NR 0, ASP 0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccGet_xc_nr_asp_info:
media class tags: NR 0, ASP 0
Nov 23 19:31:39.620: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.620: //84/B583C95F8093/CCAPI/cc_api_call_proceeding:
Interface=0x22847B14, Progress Indication=NULL(0)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_delay_xport:
CallInfo(delay xport=TRUE)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_alert:
Interface=0x22847B14, Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_alert:
Call Entry(Retry Count=0, Responsed=TRUE)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallAlert:
Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallAlert:
Call Entry(Responsed=TRUE, Alert Sent=TRUE)htsp_alert_notify
Nov 23 19:31:39.628: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_ALERT]fxols_offhook_alert
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_notify:
Data Bitmask=0x5, Interface=0x22847B14, Call Id=84
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_get_ssCTreRoutingNotSupported:
CallInfo(ssCTreRoutingNotSupported=FALSE)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_get_ccm_detected:
CallInfo(ccm detected=TRUE)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallNotify:
Data Bitmask=0x5, Call Id=83htsp_call_service_msghtsp_call_service_msg not EFXS (2)
Nov 23 19:31:39.672: //84/B583C95F8093/CCAPI/ccIsInfoRingback:
Returning dpRingBack=0
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_connected:
Interface=0x22847B14, Data Bitmask=0x1, Progress Indication=NULL(0),
Connection Handle=0
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_notify:
Data Bitmask=0x7, Interface=0x22847B14, Call Id=84
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Network, Params=0x0, Call Id=83
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
(confID=0xFFFFFFFF, callID1=0x53, gcid=B583C95F-53AC11E3-8093C8EE-BDE4256A, tag=0x0)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/ccConferenceCreate:
(confID=0xFFFFFFFF, callID2=0x54, gcid=B583C95F-53AC11E3-8093C8EE-BDE4256A, tag=0x0)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
Conference Id=0xFFFFFFFF, Call Id1=83, Call Id2=84, Tag=0x0
Nov 23 19:31:39.700: htsp_call_bridged invoked
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_bridge_done:
Conference Id=0x21, Source Interface=0x3CE27724, Source Call Id=83,
Destination Call Id=84, Disposition=0x0, Tag=0xFFFFFFFF
Nov 23 19:31:39.700: //84/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
Nov 23 19:31:39.700: cc_api_get_xcode_stream : 4819
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_bridge_done:
Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_generic_bridge_done:
Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0x21, Destination Call Id=84)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0x21, Destination Call Id=83)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
Nov 23 19:31:39.700: confID:0x21; callEntry1 callID1:0x53, type:6; callEntry2 callID2:0x54, type:1
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_caps_ind:
Destination Interface=0x22847B14, Destination Call Id=84, Source Call Id=83,
Caps(Codec=0x1, Fax Rate=0x1, Fax Version:=0, Vad=0x1,
Modem=0x2, Codec Bytes=20, Signal Type=3)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_get_ssCTreRoutingNotSupported:
CallInfo(ssCTreRoutingNotSupported=FALSE)
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_get_ccm_detected:
CallInfo(ccm detected=TRUE)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallNotify:
Data Bitmask=0x7, Call Id=83htsp_call_service_msghtsp_call_service_msg not EFXS (2)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_process_notify_bridge_done:
Conference Id=0x21, Call Id1=83, Call Id2=84
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ind:
Destination Interface=0x3CE27724, Destination Call Id=83, Source Call Id=84,
Caps(Codec=0x1, Fax Rate=0x2, Fax Version:=0, Vad=0x1,
Modem=0x0, Codec Bytes=160, Signal Type=2)
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ack:
Destination Interface=0x3CE27724, Destination Call Id=83, Source Call Id=84,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Fax Version:=0, Vad=OFF(0x1),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=9438)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_caps_ack:
Destination Interface=0x22847B14, Destination Call Id=84, Source Call Id=83,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Fax Version:=0, Vad=OFF(0x1),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=9438)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallConnect:
Progress Indication=NULL(0), Data Bitmask=0x1
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallConnect:
Call Entry(Connected=TRUE, Responsed=TRUE)
Nov 23 19:31:39.704: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_CONNECT]fxols_offhook_connect
Nov 23 19:31:39.704: htsp_timer_stop
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_voice_mode_event:
Call Id=83
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_voice_mode_event:
Call Entry(Context=0x230F9C10)
Nov 23 19:31:39.704: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice
Nov 23 19:31:39.932: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_DSP_SIG_0110]fxols_rvs_battery
Nov 23 19:31:39.932: htsp_timer_stop2
Nov 23 19:31:39.932: htsp_timer_stop2
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_call_disconnected:
Cause Value=16, Interface=0x22847B14, Call Id=84
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/ccConferenceDestroy:
Conference Id=0x21, Tag=0x0
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/ccConferenceDestroy:
Nov 23 19:31:48.860: confID:0x21; callEntry1 callID1:0x53, type:6; callEntry2 callID2:0x54, type:1
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/cc_api_bridge_drop_done:
Conference Id=0x21, Source Interface=0x3CE27724, Source Call Id=83,
Destination Call Id=84, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_bridge_drop_done:
Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/cc_generic_bridge_done:
Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/ccCallDisconnect:
Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/ccCallDisconnect:
Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/cc_api_get_transfer_info:
Transfer Number=NULL
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/ccCallDisconnect:
Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/ccCallDisconnect:
Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/cc_api_get_transfer_info:
Transfer Number=NULL
Nov 23 19:31:48.864: htsp_timer_stop3
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_get_transfer_info:
Transfer Number=NULL
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x22847B14, Tag=0x0, Call Id=84,
Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
Nov 23 19:31:48.876: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.876: :cc_free_feature_vsa freeing 3D1B9898
Nov 23 19:31:48.876: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.876: vsacount in free is 1
Nov 23 19:31:48.884: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_HTSP_RELEASE_REQ]fxols_offhook_release
Nov 23 19:31:48.884: htsp_timer_stop
Nov 23 19:31:48.884: htsp_timer_stop2
Nov 23 19:31:48.884: htsp_timer_stop3
Nov 23 19:31:48.884: [0/0/0] set signal state = 0x4 timestamp = 0
Nov 23 19:31:48.884: htsp_timer - 2000 msec
Nov 23 19:31:48.884: //83/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x3CE27724, Tag=0x0, Call Id=83,
Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Nov 23 19:31:48.884: //83/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
Nov 23 19:31:48.884: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.884: :cc_free_feature_vsa freeing 3D1B9978
Nov 23 19:31:48.884: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.884: vsacount in free is 0
Nov 23 19:31:49.156: htsp_process_event: [0/0/0, FXOLS_GUARD_OUT, E_DSP_SIG_0110]
Nov 23 19:31:50.884: htsp_process_event: [0/0/0, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
Nov 23 19:31:50.884: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_DSP_SIG_0100]
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