H323 Gateway Redundancy In CCM
HI
I would like to configure 2 h323 gateway on the cisco call manager,Each gateway's have got 2 E1's.
I would like to know is it possible to configure redundancy on the ccm ? if we can how the ccm will come to know the first gateway is down or fully utilized..?
Please advise me...
Thanks.
Nazeer
yes, it's possible, this is from a similar post gogasca replied:
The default behavior in IOS, when there are no valid POTS dial-peers for a call to go out of, is to return âunallocated numberâ as a cause code to CCM. This can happen even if a t1 goes down. When the T1 goes down, the POTS peer is marked as down, and when CCM sends a call to the gateway, it'll return a UAN cause code to the ccm, causing CCM (by default) to stop hunting for other available gateways. There are ways to change the behavior using service parameters in CCM, but this behavior didn't make sense to me, so we looked for other ways to do it.
It turns out that you can issue the global command "no dial-peer outbound status-check pots" on the IOS GW, if you're opposed to changing the CCM behavior. What this command will do is cause the dial-peer to stay up. IOS will try and route the call, and when the T1 is down, it returns "No circuit available" to CCM. When CCM receives this cause code, it knows there's been a non-user error, and continues hunting, achieving the desired behavior.
In your case since it is H323 it will be retruning only H225 release comp message with proper code to continue re-routing.
The IOS code will translate any code we get from pots side to IP side
also, if possible, please try searching within netpro first as it's very probable someone already asked the same before
HTH
javalenc
if this helps, please rate
Similar Messages
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Gateway Failover with CCM 4 1 3b
Hi
I am having issues with gateway redundancy on CCM 4-1-3b. Basically when the Primary route group gateway fails, the backup gateway does not work. You can not make external calls out of this gateway unless you change the priority of the gateways in the route group manually. The only other time it failsover is if ip connectivity fails between the primary gateway and the CCM. Has anyone had any issues like this or is there a bug with this version of CCM? Failover with RG/RL should be a basic feature with CCM?Hi Derek,
CCM regards H323 gateways as a black hole. CCM does not have the ability see the busy/fail status of the H323 ports. As you said, the only way CCM would switch to the second gateway, in the Route List, would be if IP connectivity was lost.
I would recommend that you additionally configuring outbound VOIP dial-peers (with a higher preference value) to forward the call from your first gateway to your second gateway when the first gateway is up and, for whatever reason, the outward ports are busy or down.
Regards
Chris -
Why we dont' see H323 gateway and SIP trunk "registered" in CCM?
in CCM admin, we see the status of H323 gateways and SIP trunks have no "registered to xxxx" status.
Unlike devices with other protocols, there is no registration mechanism for these.
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Caller ID not working on H323 Gateway
Hi,
I have a CCM Version 4.1.3 set-up to cater for 200 phones, I have a VG gateway in one of the sites that uses an FXO card for local calls, this VG is set-up as an H323 gateway, however calls coming throguh the PSTN line onto the FXO card and then onto an IP phone are not displaying Caller ID.
Just wondering if anyone has seen this before, I know that on an MGCP gateway the FXO cards don't support Caller-ID but we have it set-up as H323.
Thanks
PaulFunny, this is a question straight from one of the CCVP tests, although I won't say which one.
Try "caller-id enable" in global mode and "station-id name" on the voice-port.
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_configuration_example09186a00800a9a49.shtml -
Adding MGCP FXS Ports to H323 Gateways
Currently all of our Gateways are H.323 gateways. Due to a business requirement we are now going to be enforcing our users to use forced authorization codes to place LD calls. In order to facilitate this on our analog phones it seems the only option is to use MGCP gateways.
From what I understand we can run multiple signalling protocols on voice gateways. We have a variety of gateway models but by and large most of these gateways are VG224 models. I think what I would like to do is keep the current h.323 dial-peer and voice-port settings for the PLAR emergency phones that we have on these gateways and only change the analog phones to MGCP.
Most of the route patterns to these h323 gateways look like this... 102[0-5] and then the dial peers on the individual gateways route to the appropriate voice port like this...
dial-peer voice 1020 pots
huntstop
destination-pattern 1002
port 2/21
The Voice port config looks like this...
voice-port 2/21
timeouts interdigit 7
description tie pr 1520
station-id name PTRM 1020
station-id number 1020
caller-id enable
My plan is to create the MGCP Gateways in CUCM as wells as the DN's... in this example x1020. I will then enable MGCP on the gateways. After that my assumption is that I can individually remove the Voice-port and dial-peer configurations and then add the MGCP dial peers with the port and "service MGCPAPP" commands.
My other option is to redo the entire gateway at the same time and schedule after-hours down-times to make the change. I want to avoid this if possible as we have 40+ gateways that need to be changed.
Basically I just need some guidance or confirmation if my plan will work or if there is a better way to do this? Are there any caveats or known issues I should look out for when running multiple signalling protocols on the same gateway?
Thanks,
Trav MooreThanks Aaron,
I was wondering about the MGCP ccm-config command but was worried it would re-write the entire h.323 gateway to MGCP. Good to know that it won't and that this is a potential option.
I actually do prefer the idea of only having one signalling protocol (I would like to go all SIP if not for the FAC codes needed). Unfortunately any maintenance that I do that impacts end-users requires a lot of after-hours scheduling and maintenance alerts. These gateways have a combination of fax-machines, PLAR's (emergency phones and overhead paging), and analog phones. Maybe eventually I can migrate all of these ports to MGCP. For now the analog phones are the only ones that must be converted and if I can quickly convert them without anyone noticing aside from the minimal reset in CUCM then this would be ideal.
Thanks! -
CUCM and H323 gateway-Cause i = 0x80A6 - Network out of order
Hi,
I cant get calls into the CUCM from a H323 gateway. Incoming external calls here out of service message or Number Unobtainable. I've attched logs if anyone can help?
RichHi Alex,
Thanks for reply. I should've attached the full config as it shows the Translation Rules. See below.
Can anyone help?
voice translation-rule 2
rule 1 /^56/ /5\2/
rule 2 /^6/ /5/
rule 3 /2/ /52/
voice translation-rule 3
rule 1 /^1\(.........$\)/ /01\1/
rule 2 /^2\(.........$\)/ /02\1/
rule 3 /^7\(.........$\)/ /07\1/
rule 4 /^8\(.........$\)/ /08\1/
rule 5 /^4\(......$\)/ /01914\1/
rule 6 /^2\(......$\)/ /01912\1/
voice translation-rule 6
rule 1 /^[1-9]/ /90\0/ type international international
voice translation-rule 7
rule 1 /^1/ /901/
rule 2 /^2/ /902/
rule 3 /^3/ /903/
rule 4 /^4/ /904/
rule 5 /^5/ /905/
rule 6 /^6/ /906/
rule 7 /^7/ /907/
rule 8 /^8/ /908/
rule 9 /^9/ /909/
voice translation-profile INCOMING
translate calling 7
translate called 2
voice translation-profile OUTGOING
translate called 3
voice-card 0
dsp services dspfarm
interface FastEthernet0/0
ip address 192.168.178.66 255.255.255.192
ip pim dense-mode
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.178.66
interface FastEthernet0/1
no ip address
duplex auto
speed auto
no keepalive
interface BRI0/1/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
isdn incoming-voice voice
isdn static-tei 0
interface BRI0/1/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
isdn incoming-voice voice
isdn static-tei 0
control-plane
voice-port 0/1/0
translation-profile incoming INCOMING
compand-type a-law
cptone GB
description ISDN2 .......... lines 1+2
voice-port 0/1/1
translation-profile incoming INCOMING
compand-type a-law
cptone GB
description ISDN2 .......... lines 3+4
ccm-manager music-on-hold
mgcp fax t38 ecm
dial-peer cor custom
dial-peer voice 999 pots
destination-pattern 999
port 0/1/0
forward-digits all
dial-peer voice 9991 pots
destination-pattern 999
port 0/1/1
forward-digits all
dial-peer voice 112 pots
destination-pattern 9112
port 0/1/0
forward-digits 3
dial-peer voice 1121 pots
destination-pattern 9112
port 0/1/1
forward-digits 3
dial-peer voice 9999 pots
destination-pattern 9999
port 0/1/0
forward-digits 3
dial-peer voice 99991 pots
destination-pattern 9999
port 0/1/1
forward-digits 3
dial-peer voice 100 voip
preference 2
destination-pattern 52...
progress_ind setup enable 3
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.0.150
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 101 voip
preference 1
destination-pattern 52...
progress_ind setup enable 3
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.203.20
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 1 pots
translation-profile outgoing OUTGOING
preference 3
destination-pattern 0T
translate-outgoing called 10
incoming called-number .
fax rate disable
direct-inward-dial
port 0/1/0
forward-digits all
dial-peer voice 11 pots
translation-profile outgoing OUTGOING
preference 4
destination-pattern 0T
incoming called-number .
fax rate disable
direct-inward-dial
port 0/1/1
forward-digits all
dial-peer voice 9 pots
translation-profile outgoing OUTGOING
preference 1
destination-pattern 9T
incoming called-number .
fax rate disable
direct-inward-dial
port 0/1/0
forward-digits all
dial-peer voice 91 pots
translation-profile outgoing OUTGOING
preference 2
destination-pattern 9T
incoming called-number .
fax rate disable
direct-inward-dial
port 0/1/1
forward-digits all
dial-peer voice 2 pots
translation-profile outgoing OUTGOING
destination-pattern 2......
incoming called-number .
fax rate disable
direct-inward-dial
port 0/1/1
forward-digits all
dial-peer voice 21 pots
translation-profile outgoing OUTGOING
destination-pattern 2......
incoming called-number .
fax rate disable
direct-inward-dial
port 0/1/1
forward-digits all
dial-peer voice 52982 voip
preference 1
destination-pattern 562982
progress_ind setup enable 3
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.0.150
dtmf-relay h245-alphanumeric
no vad -
We have a CUCM ver 8.6.2.22900-9.
For our PSTN access we have the following scheme:
CUCM <========> Cisco2911 <======> ISP-------> PSTN
H323 tk SIP tk
Our Cisco2911 is running ver 150-1.M4.
We configure Cisco 2911 as H323 gateway on CUCM, and we established a SIP tk with our ISP.
When we tested incoming calls we received them at router but they are unable to acceess CUCM,
When we tested outgoing calls we got reorder tone and they did not reach our voice gateway.
We run "debug cch323 all" and tested with one incoming and one outgoing call, and we got the out on attached file "debug h323 forum"
CUCM Ip : 10.1.2.2
Router's H323 int: 10.2.2.1
Do you have any idea about the root cause?
Can be an access list problem preventing proper H323 traffic??
Thanks in advanced
EnriqueDo you have voip dial-peers pointing back to your cucm from your gateway? If not then the call wouldn't route onward to the cucm, but the gateway would also ignore communications from that souce because of toll fraud prevention.
http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/112083-tollfraud-ios.html -
The gatekeeper doesn't send an ACF to the h323 gateway
Hi,
I found a h323 gateway can't make any international calls thru a gatekeeper.
From the debug log, the gatekeeper doesn't send an ACF to the gateway after received a LCF from another gatekeeper.
Is it a know bug or configuration issue? Attached is the debug file.
Thanks in advance.
Rgds,
Ivan Cheng
hq01#csim start 6568465338
hq01#show log
Syslog logging: enabled (0 messages dropped, 15 messages rate-limited, 0 flushes, 0 overruns)
Console logging: level debugging, 3799 messages logged
Monitor logging: level debugging, 480 messages logged
Buffer logging: level debugging, 3813 messages logged
Logging Exception size (4096 bytes)
Count and timestamp logging messages: disabled
Trap logging: level notifications, 583 message lines logged
Logging to 155.161.76.72, 583 message lines logged
Logging to 155.161.135.135, 583 message lines logged
Logging to 192.170.75.40, 583 message lines logged
Log Buffer (4096 bytes):
00 44004200 34004500 31003800 30003000 30003000 30003000 33003900 00000000 00000000 00000000 00000000 FA232100 11000000 00000000 00000000 00000000 00000100
Apr 3 14:08:20 GMT:
Apr 3 14:08:20 GMT: csim_do_test: cid(475), ev(11), disp(0)
Apr 3 14:08:20 GMT: csimTraceSct: cid(475),st(0),oldst(0)
Apr 3 14:08:20 GMT: csim err csimDisconnected recvd DISC cid(475)
Apr 3 14:08:20 GMT: //475/xxxxxxxxxxxx/CCAPI/ccCallDisconnect: (callID=0x1DB, cause=0x3F tag=0x0)
Apr 3 14:08:20 GMT: //475/xxxxxxxxxxxx/CCAPI/ccCallDisconnect: calling accounting start for callID=475 leg_type=0
Apr 3 14:08:20 GMT: //475/xxxxxxxxxxxx/CCAPI/ccCallDisconnect: existing_cause = 0x3F, new_cause = 0x3F
Apr 3 14:08:20 GMT: //475/xxxxxxxxxxxx/CCAPI/ccCallDisconnect: using the existing_cause 0x3F
Apr 3 14:08:20 GMT: //475/xxxxxxxxxxxx/CCAPI/cc_api_get_transfer_info: (callID=0x1DB)
Apr 3 14:08:20 GMT: //475/xxxxxxxxxxxx/CCAPI/ccTDUtilGetDataByValue: CallID[475], tagID[30], instID[-1]
Apr 3 14:08:20 GMT: //475/xxxxxxxxxxxx/CCAPI/ccTDPvtProfileTableObjectAccessManager: No profileTable set for callID[475]
Apr 3 14:08:20 GMT: //475/xxxxxxxxxxxx/CCAPI/ccTDUtilGetDataByValue: No tdObject found in profileTable for tagID[30] of callID[475]
Apr 3 14:08:20 GMT: H.225 SM: process event H225_EVENT_RELEASE, for callID 1DBcch323_run_h225_sm: received event H225_EVENT_RELEASE while at state H225_WAIT_FOR_RAS_CONF
cch323_run_h225_sm: Unexpected event[9] in CCH323_H225_STATE_WAIT_FOR_RAS_CONF state
Apr 3 14:08:20 GMT: H.225 SM: changing from H225_WAIT_FOR_RAS_CONF state to H225_WAIT_FOR_DRQ state for callID 1DB
Apr 3 14:08:20 GMT: RAS INCOMING ENCODE BUFFER::= 40 1EA8
Apr 3 14:08:20 GMT:
Apr 3 14:08:20 GMT: RAS INCOMING PDU ::=
value RasMessage ::= disengageConfirm :
requestSeqNum 7849
Apr 3 14:08:20 GMT: H.225 SM: process event H225_EVENT_RAS_SUCCESS, for callID 1DBcch323_run_h225_sm: received event H225_EVENT_RAS_SUCCESS while at state H225_WAIT_FOR_DRQ
Apr 3 14:08:20 GMT: H.225 SM: changing from H225_WAIT_FOR_DRQ state to H225_IDLE state for callID 1DB
Apr 3 14:08:20 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_api_icpif: expect factor = 0
Apr 3 14:08:20 GMT: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2, ConnectionId 0 0 0 0, SetupTime 14:08:04.217 GMT Thu Apr 3 2008, PeerAddress 6568465338, PeerSubAddress , DisconnectCause 65 , DisconnectText message in incomp call state (101), ConnectTime 14:08:20.637 GMT Thu Apr 3 2008, DisconnectTime 14:08:20.637 GMT Thu Apr 3 2008, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0Hi Brandon,
Here you are.
Thanks in advance.
Rgds,
Ivan Cheng -
HSRP For the Gateway Redundancy.
Hi all
i just need a simple how to configure 2 Routers(R1;R2) to run HSRP For the Gateway Redundancy ,if one of the 2 routers Fail.should i connect the 2 routers 2gather via cross cable.than one straight cable to the 2 separate distribution switch.(2 Etherchanel configured between Dist switch)PS LIST ur optimum Configuration
Ur help very much AppreciatedHi,
i think you talking about campus network where you have two distribution two access and two core router's.
With that prospects.
My suggesion will be to have etherchannel between distribution switches and both the distribution switch should be connected to both the core router.
than use HSRP in distribution swithes.
configuration and diagram is given below.
do let us know if you want any more information.
interface FastEthernet2
ip address 172.69.90.1 255.255.255.0
standby priority 200
standby preempt
standby ip 172.69.90.6
interface FastEthernet3
ip address 172.69.91.1 255.255.255.0
standby priority 200
standby preempt
standby ip 172.69.91.6
like the above configuration you can configure second switch also you can apply on vlan interface too.
HTH -
Gatekeeper showing priority 0 on h323 gateway
Hello,
I have an H323 gateway configured with primary and alternate gatekeepers. Priorities were not configured, so both gatekeepers SHOULD show as 127. However, the first gatekeeper configured shows a priority of 127, the other gatekeeper shows a priority of 0.
I am able to connect to both gatekeepers from the gateway.
I did some research and see that there are some bugs out there on earlier versions which cause the priority to show incorrectly. but i'm wondering if the bug would only be cosmetic (show the priority as incorrect) or if the priority is actually incorrect.
This is happening on multiple gateway / gatekeeper configurations all with the same gatekeeper priority default configuration and showing the same 0 / 127 priorities in the show gateway
I am wondering if the registration failure alerts is due to the priority showing as 0, and if the fix might be to manually configure the gatekeepers' priority.
Has anyone else seen this issue?
H.323 ITU-T Version: 4.0 H323 Stack Version: 0.1
H.323 service is up
Gateway GATEWAY is registered to Gatekeeper GATEKEEPER-PRI
Alias list (CLI configured)
H323-ID GATEWAY
Alias list (last RCF)
H323-ID GATEWAY
H323 resource thresholding is Disabled
Permanent Alternate Gatekeeper List
priority 0 id GATEKEEPER-ALT ipaddr 10.76.155.172 1719 register needed
priority 127 id GATEKEEPER-PRI ipaddr 10.76.155.171 1719 register needed
Primary gatekeeper ID GATEKEEPER-PRI ipaddr 10.76.155.171 1719
Showing alerts in logs:
074975: Mar 14 10:45:51.195 EDT: %CCH323-2-GTWY_REGSTR_FAILED_ALT_GK: Gateway GATEWAY failed attempt to register with Alternate Gatekeeper GATEKEEPER-ALT
074976: Mar 14 10:45:51.251 EDT: %CCH323-6-REGSTR: Gateway GATEWAY registered with Gatekeeper GATEKEEPER-PRI
074994: Mar 14 10:46:06.715 EDT: %CCH323-2-GTWY_REGSTR_FAILED_ALT_GK: Gateway GATEWAY failed attempt to register with Alternate Gatekeeper GATEKEEPER-ALT
074995: Mar 14 10:46:06.771 EDT: %CCH323-6-REGSTR: Gateway GATEWAY registered with Gatekeeper GATEKEEPER-PRI
Configuration:
interface Loopback0
ip address 10.174.102.171 255.255.255.255
ip pim sparse-dense-mode
h323-gateway voip interface
h323-gateway voip id GATEKEEPER-PRI ipaddr 10.176.155.171 1719
h323-gateway voip id GATEKEEPER-ALT ipaddr 10.176.155.172 1719
h323-gateway voip h323-id GATEWAY
h323-gateway voip tech-prefix 700#
h323-gateway voip bind srcaddr 10.174.102.171
Thank you!BTW - my router is a 3845, IOS version: 12.4(15)T9
-
Help on set up branch office with 2921 H323 gateway
I setup a new branch office with 2921 H323 gateway and cucm in HQ. When I call a number in remote office, I get dead silence and busy tone. However, user can hear ring at the remote location and able to answer the phone. I was able to talk to him. Any place I need to check?
Question #2, should cucm in HQ handle all calls between HQ and remote office? I tried to call from my VoIP phone to remote office VoIP phone and monitored remote office GW running "debug voip ccapi inout". I saw messages like gateway is handling calls. Is this normal?
I'm fairly new to VoIP environment, still trying to learn. Thanks. Let me know if you need anything to troubleshoot this.I think that's where I'm confused. I'd like to have CUCM to handle every calls for remote sites. when I searched for the number I'm dialing for remote office, there is a route pattern that covers this number and it points to gateway. This route pattern covers all of their local numbers including our remote office numbers. For example, I have 9.1201456XXXX point to H323 GW. That I got it. However, I want the numbers belong to our office like 1111 don't go to GW for call processing. Do I make sense? I want only their local call to remote office go through GW not our internal call between our offices. I'm sorry if I don't make much sense. Thanks for your help.
-
Can not add H323 gateway again
Hi all,
I am doing lab with CUCM 7.1. So, I added one h323 gateway (10.15.242.59) and then I deleted it. Now, I add new h323 gateway (same Ip address 10.15.242.59), one message displays:
"Add failed. One of the required fields on the page has the same value as an entry that already exists in the database. Please check the corresponding Find List page to verify your entry does not exist."
But I also check in route group and gateway list, there are no one which has this IP address.
Thanks,
ThucHi.
You can also check if device still exist in the DB by issuing:
run sql select * from device where name='10.15.242.59'
and see if you obtain some result.
HTH
Regards
Carlo
Please rate all helpful posts
"The more you help the more you learn" -
Callmanager 3.3(5), H323 gateways not rolling over
Outbound calls are not overflowing from one PRI on one h323 gateway to another. Once all 23 channels are used, I get a busy tone.
I have two h323 gateways. Both are cisco 3725 routers and each has a PRI on it. Each H323 is in theri own route group. And the Route-List contains only these two Route-groups. So it's a pure h323 route-list.
I have tried reversing the order to verify that outbound calls work on both PRI and they do.
I opened a case with cisco and mentioned changing the service parameter of "Stop routing on user busy flag" from true to false. We did that and it's still not working.
Wondering if anyone is having the same experience or have any ideas.This looks like a pretty old post but this has bitten me once before and there is a fix. In CUCM Service Parameters under Cisco Callmanager Active/ Clusterwide Parameters/Route Plan - Change
"Stop Routing on Unallocated Number Flag" and "Stop Routing on User Busy Flag" to False. This will allow H323 gateways and SIP Trunks to advance to the next Gateway or Trunk in the Route Group/Route List. If it is set to True the backup Gateways and or SIP Trunks will never be used.
Thanks. -
H323 Gateway Intermittent MOH Silence
We are using a 3925 router - c3900-universalk9-mz.SPA.150-1.M4.bin - as an H323 gateway with 2 PRI's. We have a PUB and a SUB running -
System version: 8.5.1.11900-21 When we place people on hold we get MOH very intermittently. Out of 10 calls 8 might get MOH and 2 get silence. I have checked codecs (all calls G711), I have restarted IPMediaStreaming. The weird thing is that some calls work and some do not. We cannot isolate any factors to be the same or possible causes. I uploaded the voice portion of the router to see if it helps. Thanks for any help.
Thanks,
DonnieHEllo,
I had the same issue. Check the BUG ID. My IOS version is 15.0(1)M4 on a Cisco 2911. The affected IOS version described on the Bug ID.
I've made the Option2 Workaround and the issue got solved.
Thanks for the info.
Rui -
Hello All, can someone tell me how Lync 2013 handles PSTN gateway redundancy? I currently have two cisco gateways connected to my Lync 2013 environment. It appears that under normal operation, Lync just round robins between the two gateways. If I purposely
take one of the gateways down, almost every other outgoing Lync call will fail. Do I have to do something within Lync so that it can detect when a gateway is unavailable and use the surviving one?
Also, how can I make sure that Lync will use the other gateway if the first one is out of T1 channels.That is probably the explanation. If you want to research further you will have to look at what error the gateway is providing Lync. In all likelihood when the call is failing there will be some type of error that points to an ISDN error. For example,
in trunk busy situation Lync will get a 503 error with an additional reason (e.g. msgwsip-reason="Q.850; cause=34";). The cause shown here is a the ISDN error code. It can
take some time for Lync to mark these gateways as down.
What return error would make lync use the other gateway? By default, my gw is responding with a SIP 404 error: SIP/2.0 404 Not Found. I could however make it respond with a 503 error: SIP/2.0 503 Service Unavailable.
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.86.176.110:5060;branch=z9hG4bK647452a554f9
From: <sip:[email protected]>;tag=114175~98718eca-fc35-493e-8cde-1d27b61c679c-30156882
To: <sip:[email protected]>;tag=1B49687C-261B
Date: Tue, 08 Jan 2013 22:54:54 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-15.2.3.T
Reason: Q.850;cause=1
Content-Length: 0
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.86.176.110:5060;branch=z9hG4bK6473356b69f7
From: <sip:[email protected]>;tag=114171~98718eca-fc35-493e-8cde-1d27b61c679c-30156879
To: <sip:[email protected]>;tag=1B4927B4-285
Date: Tue, 08 Jan 2013 22:54:37 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-15.2.3.T
Reason: Q.850;cause=34
Content-Length: 0
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