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Outgoing Calls to mobile number using sip trunk

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WireShark log shows that no outgoing calls are working, regardless of whether they are made to an external number, or to a CUCM extension.

Outgoing calls to local CUCM extensions were working beforehand - as per this support forum thread.

What has changed that stopped outgoing calls to local extensions working?

Traces show that the calls are now made form a different extension: 782 instead of 751, so maybe some configuration issue on CUCM that is ignoring calls from 782 ?

Once you have outgoing calls to local extensions working again then you can try making calls to external numbers. Again, the Cisco Call Manager would be responsible for routing those calls correctly.

ws_outgoing_calls.png

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I would like to know is there any possibility to call the mobile number using .Net application?

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VoiceGuide APIs can be called using .NET's WCF protocol.

So you can load outgoing calls into VoiceGuide from a .NET application by calling the Dialer_OutDialQueAdd API function through WCF.

.NET application can also load calls using other approaches:

- Saving list of numbers to be called in an Out Dial File, which VoiceGuide reads in automatically.
- Adding dial entries directly into the VoiceGuide OutDial Database.

Please see:

https://www.voiceguide.com/vghelp/source/html/outbound-ivr-dialer-introduction.htm

https://www.voiceguide.com/vghelp/source/html/dial_vgdb_external_config.htm

https://www.voiceguide.com/vghelp/source/html/com_dialer_outdialqueadd.htm

 

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Need your help regarding  CUCM  Configuration for sip trunk because not able to call mobile number please provide any link or remotely support 

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Are the calls to local extensions working again now?

As mentioned previously:

Outgoing calls to local CUCM extensions were working beforehand - as per this support forum thread.

Once calls to local extensions work again then you can start working on getting calls to external mobiles working.

 

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Yes call to local extension is working but not working for mobiles.Any configuration is needed

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1 hour ago, SupportTeam said:

Outgoing calls to local CUCM extensions were working beforehand - as per this support forum thread.

this tread is related to transfer i will work on this once i get the license because two lines should be available for that but now i am working on outgoing call to mobile and international numbers for that any configuration needed in voice guide?

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The CUCM configuration needs to be looked to to see why CUCM is ringing the local extensions and connecting call, but not making connections when it is instructed to call an outside number instead.

CUCM logs may show you why CUCM is not making connections to outside numbers.

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ok but any support of CUCM from your end.Furthermore,I am just confirming there is no need of configuration in voice guide if using sip trunk for outgoing calls to local mobile and international mobile

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This support forum does not extend to debugging CUCM configuration and settings.

Outgoing call requests to all numbers are all made in same way by VoiceGuide. There is no separate configuration required in VoiceGuide to call different numbers.

For some reason your CUCM does not currently make the outgoing call when an external number is specified. You would need to see how CUCM handled the call request to see why CUCM did not proceed with placing that call.

If CUCM is responding to SIP INVITE then the reason for not progressing with call to external number could even be included in the response. Otherwise suggest looking at internal CUCM logs or having a closer look at all the CUCM configurations which would affect the placing of the external call.

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Calling on my mobile No using outbound IVR once i received a call i did not answer a call as i rejected a call but line status monitor still playing the script.If i answer the call then only it should start playing the script.

same problem after i answer the call and ended the call from my end but line status monitor still playing the script.Once i ended the line status monitor should hang up the call immediately.

LineStatusMonitor.PNG

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Sounds like you are able to now call external mobile numbers.

Most likely what is happening is that when you press "Reject" on your mobile phone the call is just transferred to your voicemail, not disconnected.

You can use WireShark to confirm whether the phone company actually disconnected the call or not. You would probably want to capture the traffic between your PBX (CallManager) and the phone company to confirm if any call state messaging is even sent by your phone company when you press "Reject" on your mobile phone.

 

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same problem after i answer the call and ended the call from my end but line status monitor still playing the script.Once i ended the line status monitor should hang up the call immediately.

When called person hangs up the call on a mobile phone then you would expect to receive advice that call was disconnected.

But if you dial out to a landline then many phone network will not advise that remote end hung up the call. The IVR script just needs to rely on timing out when awaiting response from called person and hang up.

Maybe your phone company does not send any messaging when the mobile phone that received the outgoing call hangs up. You would need to perform WireShark tracing on what messages are sent by your phone company when the remote person hangs up.

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Dear Voice Guide,

Step 1:

Outgoing call using Outbound loader did not work.

Step 2:

Receive a incoming  call to voice guide working

Step 3:

Outgoing call using Outbound loader Working but after few minutes i try to call again the same as step 1.Please check the attached log 

Logs.zip

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WireShark shows that VoiceGuide tried making the first outgoing call, but did not receive any reply to the repeated SIP INVITE's sent to IP 10.99.10.31

You need to look into why 10.99.10.31 did not reply at all to those multiple SIP INVITE's.

Perhaps the networking is not set up in way that ensures constant SIP connectivity between 172.16.10.22 and 10.99.10.31, and the outgoing INVITEs of that first call are lost along the way? But when a SIP message is sent from 10.99.10.31 to 172.16.10.22 the routes are then temporarily re-established?

 

image.thumb.png.a29a8cbd8847231f2a7824c8ca3f6e24.png the

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But All setup is done in CU-CM for outgoing and incoming not able to find the exact issue whether  this is issue from  from Voice Guide or from CU-CM

 My question is why outgoing is working after incoming call .

 

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VoiceGuide/HMP sends out the outgoing call INVITE in same way in both cases.

But no reply can be seen in first call.

So the question is why no reply arrives back to VoiceGuide/HMP.

You have to establish why the side that should be replying (10.99.10.31) is not replying.

There is a high chance that you will find that as advised before, that: "networking is not set up in way that ensures constant SIP connectivity between 172.16.10.22 and 10.99.10.31" and that "when a SIP message is sent from 10.99.10.31 to 172.16.10.22 the routes are then temporarily re-established".

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I want you to check this Remotely because we not able to find this.

1 minute ago, SupportTeam said:

networking is s not set up in way that ensures constant SIP connectivity between 172.16.10.22 and 10.99.10.31" and that "when a SIP message is sent from 10.99.10.31 to 172.16.10.22 the routes are then temporarily re-established

 

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You need to speak to people who are responsible for network connectivity between 172.16.10.22 and 10.99.10.31. They will then be able to perform necessary tracing and check configuration of equipment that is used in that network path.

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I already spoke to them they are saying everything is done from their end i would like to request you to check remotely .

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Engaging in process of consulting with your network administrators as to why the SIP messages are not traversing their network is something that can be done, but would be outside of this free Support Forum.

Please contact sales@voiceguide.com for details of Direct Support Plans which can be put in place.

Once a Direct Support Plan is in place then our support staff can remotely log into your systems (after you set up appropriate remote access software) and assist in debugging the network connectivity.

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Dear VoiceGuide,

Check below response of CUCM logs.

I checked the logs, but I cannot see any calls coming from the IVR towards CUCM.

 

I also checked the captures taken from CUCM publisher. However no SIP messages are being received by the IVR:

picture?folder=default0%2FINBOX&id=6655165916663263308&uid=image003.png%4001D4BECA.B3727140

So the invites that are sent by the IVR are not being received by CUCM. This is why we see IVR sending multiple invites to establish the call.

 

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So the invites that are sent by the IVR are not being received by CUCM. This is why we see IVR sending multiple invites to establish the call.

This confirms that networking is not set up in way that ensures constant SIP connectivity from 172.16.10.22 to 10.99.10.31, and the outgoing INVITEs from 172.16.10.22 of that first call are lost along the way.

You need to speak to people who are responsible for network connectivity between 172.16.10.22 and 10.99.10.31. They will then be able to perform necessary tracing and check configuration of equipment that is used in that network path.

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I already ask them they said Everything done from their side 

Also i would like to know what ports voice guide is sending for outgoing call 

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Quote

  they said Everything done from their side 

Well obviously it's not, because the outgoing INVITE packets are not being received by 10.99.10.31

 

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what ports voice guide is sending for outgoing call 

5060 (UDP)

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Dear Voice,

I make an outgoing call to mobile number accept the call after 10 sec i end the call from mobile but line status monitor still not hanging up.

Once i end the call from my mobile phone it should automatically end in Voice Guide  also.

Please check the logs.

Logs.zip

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WireShark trace shows that called destination number was found by and started ringing about 3 seconds after dialing, and then was answered about 3 seconds later.

Call was ended by VoiceGuide about 28 seconds after the call was answered, presumably because the VoiceGuide script timed put awaiting caller input at some stage. (vgEngine trace was not supplied so not possible to see why VoiceGuide hung up the call)

If the destination number hung up earlier then there was no SIP signaling of any sort that advised VoiceGuide that the call was ended. You should contact whoever supplies the trunks over which this outgoing call was made and advise them that they did not send any signalling indicating end of call.

If that service provider played a disconnect tone the it is possible to set up VoiceGuide to detect the disconnect tone and hang up upon hearing to the disconnect tone. Please see here for more information on how to configure the disconnect tone detection: https://www.voiceguide.com/vghelp/source/html/disconnectiondetect.htm

But the service provider should really be sending the disconnect message if the mobile phone hung up. (if you are calling a traditional analog phone the you will usually not get an explicit disconnect message, and you need to rely on tone detection and timeouts to end call, but when calling mobile phones it is expected that service provider sends disconnect message).

image.thumb.png.fdf1f91354060760a5670ab9cab20aee.png

 

 

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Please check the Vg Engine Trace and let me solution which can be configured in voice guide.

and also how to find out disconnected tone played while hang up the call from Mobile.

As of my thinking Voice Guide should handle this scenario.

6 minutes ago, SupportTeam said:

(vgEngine trace was not supplied so not possible to see why VoiceGuide hung up the call)

 

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how to find out disconnected tone played while hang up the call from Mobile.

Make an outgoing call and use a Record module in the script that runs when the outgoing call is answered. (simplest to just use script with a single Record module in it)

When the Record module starts recording hang up the mobile. The recording will now include what is heard by the system on the line after the dialed party hangs up.

Play the recorded file in any sound editor to hear if any tones were sent by Telco after the dialed party hung up. If tones are present then you can analyze them using Audacity or similar and then use ConfigLine.xml to set up VoiceGuide/HMP to detect that tone in future.

Instructions are provided here: https://www.voiceguide.com/vghelp/source/html/disconnectiondetect.htm

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Two things:

 

1. Talk to your SIP trunk provider to see whey they aren't sending a SIP 'BYE' when the called mobile hangs up.

Your SIP trunk provider needs to look at their setup. Telephone networks know when a mobile hangs up, and this should be propagated back to you. You may want to look into using a different SIP trunk provider that has their phone network interconnectivity properly set up.

 

2. Is this is really affecting your solution that much?

The call to mobile is the outgoing leg of the bridged/conference transfer, yes? That outgoing leg will be hung up by VoiceGuide when the other (incoming) leg of the connected call hangs up. Isn't this adequate in your case?

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