Outgoing calls on a CUCM SIP Trunk

[Guest mohammedzoheb]

Hi i am using Transfer to call please check the below image and logs.When press 1 call is hanging up not able to transfer please let me know how could i use transfer to call as i am using SIP Trunk. 

WireSharkLog.pcapng

1226_CallEvents.txt

[Guest mohammedzoheb]

I am using Outbound Call Loader wireshark logs show that its calling to my number but  i am not receiving a call please check .

WireSharkLog.pcapng

1226_CallEvents.txt

[SupportTeam]

On SIP Trunks you will usually need to use the "Dialout and Conference" type transfers.

Regarding the outgoing calls: WireShark shows that the outgoing INVITE sip messages are sent out, but there is no response of any kind from 10.99.10.31

Suggest checking logs on the 10.99.10.31 device side to see how the arriving SIP invite messages were processed by 10.99.10.31

[Guest mohammedzoheb]

13 minutes ago, SupportTeam said:

Suggest checking logs on the 10.99.10.31 device side to see how the arriving SIP invite messages were processed by 10.99.10.31

you mean i should check with Cisco Call Manager or can you elaborate please 

[Guest mohammedzoheb]

16 minutes ago, SupportTeam said:

On SIP Trunks you will usually need to use the "Dialout and Conference" type transfers.

Request you to provide any documentation regarding this and also is there any configuration needs to be done in config file let me know

[SupportTeam]

Quote

you mean i should check with Cisco Call Manager

Yes. SIP INVITE was sent to Cisco Call Manager, but there was no reply from CUCM.

No special configuration needed, as you are receiving the calls from that SIP trunk into VoiceGuide, as per posts in other thread.

(you are still receiving calls from that CUCM into VoiceGuide, right?)

 

[Guest mohammedzoheb]

Hi Please check the call event and steps as i am using sip trunk you ask to use Dialout and Conference but after following the step not able to transfer please check the log

1226_CallEvents.txt

1226_vgService.txt

[SupportTeam]

The "Dial and Conference" uses another line to place the outgoing call.

Right now your test system only had one HMP line, and it is used by the incoming call.

This is why you see this message in VoiceGuide logs: "No free line for 2nd leg of call"

You can obtain a temporary 2 line HMP license from here:  https://www.dialogic.com/hmp-software/hmp-license

Once you have a 2 line license you can attempt the "Dial and Conference" transfers.

Recommend confirming that normal outbound calls work first. You can test outbound calls with your current 1-line license. The second leg of the "Dial and Conference" transfer call is an outgoing call.

You may need to set the <CallerId> option on the outgoing calls, but this is usually only required if the PBX needs to authenticate outgoing calls made though the SIP trunk first.

 

CallEvent log shows:

{ "call": {
  "crn": "8000001",
  "port": "1",
  "direction": "in",
  "time_start": "2018-12-26 17:15:23",
  "events": "
    171523.186|callstate|GCEV_OFFERED|741@10.99.10.31||782@172.16.10.222|[SIP_HDR_Request_URI]{sip:782@172.16.10.222:5060}[SIP_HDR_Contact_URI]{sip:741@10.99.10.31:5060}[SIP_HDR_FROM_DISPLAY]{"zohebuddin"}[SIP_HDR_EXPIRES]{180}[SIP_HDR_CALLID]{f23e7080-c2318d6e-879fd-1f0a630a@10.99.10.31}[SIP_HDR_FROM]{"zohebuddin"<sip:741@10.99.10.31>;tag=3440826~4c075c6d-5a3b-4906-9ab9-f8e1d805b877-29066501}[SIP_HDR_TO]{<sip:782@172.16.10.222>}[SIP_Header_Via]{SIP/2.0/UDP 10.99.10.31:5060;branch=z9hG4bK9356b52fc0ef1}[SIP_Header_From]{"zohebuddin"<sip:741@10.99.10.31>;tag=3440826~4c075c6d-5a3b-4906-9ab9-f8e1d805b877-29066501}[SIP_Header_To]{<sip:782@172.16.10.222>}[SIP_Header_Contact]{<sip:741@10.99.10.31:5060>;+u.sip!devicename.ccm.cisco.com="SEPB4A8B9E85EF3"}[SIP_Header_Call-ID]{f23e7080-c2318d6e-879fd-1f0a630a@10.99.10.31}[SIP_Header_User-Agent]{Cisco-CUCM11.5}||0|2|8|0
    171523.187|event|GCEV_OFFERED|741@10.99.10.31||782@172.16.10.222|[SIP_HDR_Request_URI]{sip:782@172.16.10.222:5060}[SIP_HDR_Contact_URI]{sip:741@10.99.10.31:5060}[SIP_HDR_FROM_DISPLAY]{"zohebuddin"}[SIP_HDR_EXPIRES]{180}[SIP_HDR_CALLID]{f23e7080-c2318d6e-879fd-1f0a630a@10.99.10.31}[SIP_HDR_FROM]{"zohebuddin"<sip:741@10.99.10.31>;tag=3440826~4c075c6d-5a3b-4906-9ab9-f8e1d805b877-29066501}[SIP_HDR_TO]{<sip:782@172.16.10.222>}[SIP_Header_Via]{SIP/2.0/UDP 10.99.10.31:5060;branch=z9hG4bK9356b52fc0ef1}[SIP_Header_From]{"zohebuddin"<sip:741@10.99.10.31>;tag=3440826~4c075c6d-5a3b-4906-9ab9-f8e1d805b877-29066501}[SIP_Header_To]{<sip:782@172.16.10.222>}[SIP_Header_Contact]{<sip:741@10.99.10.31:5060>;+u.sip!devicename.ccm.cisco.com="SEPB4A8B9E85EF3"}[SIP_Header_Call-ID]{f23e7080-c2318d6e-879fd-1f0a630a@10.99.10.31}[SIP_Header_User-Agent]{Cisco-CUCM11.5}||2|0|8|0
    171523.187|command|cmd_AnswerCall|||||0|0|0|0
    171523.256|event|GCEV_ANSWERED|||||2050|0|0|0
    171523.256|callstate|GCEV_ANSWERED|||||0|256|4|0
    171523.257|state|[Welcome] Playing wav (C:\zoheb\Eng2.wav)|||||0|0|0|0
    171523.257|event|GCEV_ANSWERED|||||256|1|4|0
    171526.833|state|[Web Service 8] Web Service|||||0|0|0|0
    171526.834|command|cmd_PlayStop|||||0|0|0|0
    171526.834|dtmf||1||||49|0|0|0
    171527.574|state|[CustomerService] Playing wav (C:\zoheb\CustomerService.wav)|||||0|0|0|0
    171528.848|state|[Transfer Call] Monitored Dial and Connect to 751@10.99.10.31|||||0|0|0|0
    171528.850|state|Hanging up... [No free line for 2nd leg of call]|||||0|0|0|0
    171528.852|command|cmd_DropCall|||||0|0|0|0
    171528.876|callstate|GCEV_DROPCALL|||||0|1|32|0
    171528.891|event|GCEV_DROPCALL|||||1|0|32|0
    171528.908|callstate|GCEV_RELEASECALL|GCST_NULL||||2137|0|0|0
  ",
  "length": "5.722"
}},

 

[Guest mohammedzoheb]

Please check is there any configuration changes needs to done because  some calls i am able to call using Outbound IVR but sometimes its not able to call.

Also  i configure the script when the call is answer by live person but after answering the call after 30 to 40  sec my script is playing please check the log .

1227_CallEvents.txt

1227_1223_vgDialListLoad.txt

1227_vgService.txt

1227_ktTel.txt

WireSharkLog.pcapng

[SupportTeam]

The last 3 outgoing calls in the logs were all successful, but it looks like you have the specified a script in both the "Live Person Answer" and the "Answering Machine Answer" fields, so the system is trying to determine what answered the call and start the appropriate script.

To determine what answered the call the system waits to hear what is said/played by the side that answered the call.

It looks like on the last 3 calls nothing was said by the party answering the call, and the system is just waiting to hear something. That is why you wait o long until system eventually tomes out and starts playing the default "Live Person / Human" script.

When answering calls please say "Hello" or similar, like normal calls would be answered. You should then see system detect answer type and start script.

You can also set the "Answering Machine Answer" field to:

disable

This way the "Live Person Answer" script will be started immediately when the call is "picked up", without waiting for call recipient to say anything.

Please also see: https://www.voiceguide.com/vghelp/source/html/detectcallanswer.htm

 

122700.958|command|MakeCall_Completed|741@10.99.10.31|751@10.99.10.31||<result>ok</result><crn>134217729</crn><crnx>8000001</crnx>|0|0|0|0
122706.735|callstate|GCEV_CONNECTED|||||0|256|4|0
122706.741|state|Dialing (auto) 741@10.99.10.31 ..., doing answer detection... (DX_PAMDOPTEN)|||||0|0|0|0
122746.767|state|Human answer. Start [C:\VoiceGuideScript\play6.vgs]|||||0|0|0|0
122746.775|state|[Welcome] Playing wav (C:\zoheb\Eng2.wav)|||||0|0|0|0

123000.969|command|MakeCall_Completed|741@10.99.10.31|751@10.99.10.31||<result>ok</result><crn>134217729</crn><crnx>8000001</crnx>|0|0|0|0
123004.066|callstate|GCEV_CONNECTED|||||0|256|4|0
123004.068|state|Dialing (auto) 741@10.99.10.31 ..., doing answer detection... (DX_PAMDOPTEN)|||||0|0|0|0
123044.094|state|Human answer. Start [C:\VoiceGuideScript\play6.vgs]|||||0|0|0|0
123044.096|state|[Welcome] Playing wav (C:\zoheb\Eng2.wav)|||||0|0|0|0

122700.958|command|MakeCall_Completed|741@10.99.10.31|751@10.99.10.31||<result>ok</result><crn>134217729</crn><crnx>8000001</crnx>|0|0|0|0
122706.735|callstate|GCEV_CONNECTED|||||0|256|4|0
122706.741|state|Dialing (auto) 741@10.99.10.31 ..., doing answer detection... (DX_PAMDOPTEN)|||||0|0|0|0
122746.767|state|Human answer. Start [C:\VoiceGuideScript\play6.vgs]|||||0|0|0|0
122746.775|state|[Welcome] Playing wav (C:\zoheb\Eng2.wav)|||||0|0|0|0

 

[Guest mohammedzoheb]

6 minutes ago, SupportTeam said:

You can also set the "Answering Machine Answer" field to:

disable

yes i followed this now its working fine thanks for your support.