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New Member

Transfering calls with a phone behind a CUBE.

Hello,

My setup is as follows.

I have two softswitches, one behind my CUBE (from now on I call it "M") and another one outside the CUBE (from now on I call it "A").

I can call from A to M and viceversa. 

If a phone from A orginates a call to M, that phone from M can transfer that call to another phone from M..but I am not able to transfer that call to a phone from A (that is, a connection between two phones from A)

Softswitch M is a softswitch we are currently testing, and using a CUBE is the only way we can hook it to the network without making lots of changes in the routing tables.

I was wondering if the transfer I am not being able to complete, can be accomplished.

I will be happy to provide you with any info you require to help me with my problem.

Thank you in advance for your help.

  • IP Telephony
Everyone's tags (2)
1 ACCEPTED SOLUTION

Accepted Solutions

when the softswitch M puts

when the softswitch M puts the call of ph-A1 on hold, softswitch-M will send out the INVITE message to CUBE and the sdp of that message should have c=0.0.0.0 or a=sendonly. This is indication to CUBE that the call was put on hold.

//Suresh Please rate all the useful posts.
9 REPLIES
Bronze

Does softswitch A support sip

Does softswitch A support sip refer?

New Member

Softswitch A is an Asterisk 1

Softswitch A is an Asterisk 1.8, so it does.

This is the initial invite, the one that phone registered in A sends to CUBE.

INVITE sip:7422@10.0.6.254 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.4:5060;branch=z9hG4bK265d780a
Max-Forwards: 70
From: "Gabriel Lema" <sip:3122@10.0.6.4>;tag=as54f712b3
To: <sip:7422@10.0.6.254>
Contact: <sip:3122@10.0.6.4:5060>
Call-ID: 79ad36b919efe27361ab02ed136aafa9@10.0.6.4:5060
CSeq: 102 INVITE
Date: Thu, 12 Jun 2014 12:15:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 297

v=0
o=root 884768917 884768917 IN IP4 10.0.6.4
s=Asterisk PBX 1.8.7.0
c=IN IP4 10.0.6.4
t=0 0
m=audio 10598 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

And this is the one that CUBE sends to M.

INVITE sip:7422@10.0.10.171:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.254:5060;branch=z9hG4bK6DBF1889
Remote-Party-ID: "Gabriel Lema" <sip:3122@10.0.10.254>;party=calling;screen=no;privacy=off
From: "Gabriel Lema" <sip:3122@10.0.10.254>;tag=E193EFFC-60F
To: <sip:7422@10.0.10.171>
Date: Thu, 12 Jun 2014 12:15:09 GMT
Call-ID: 84452AA-F16211E3-AEF195C0-9B8997B@10.0.10.254
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0138615386-4049736163-2934674880-0163092859
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1402575309
Contact: <sip:3122@10.0.10.254:5060>
Call-Info: <sip:10.0.10.254:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 280

v=0
o=CiscoSystemsSIP-GW-UserAgent 2759 7351 IN IP4 10.0.10.254
s=SIP Call
c=IN IP4 10.0.10.254
t=0 0
m=audio 19398 RTP/AVP 8 0 101 19
c=IN IP4 10.0.10.254
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000

Thanks,

Gabriel

could you please collect

could you please collect "debug ccsip all " & debug voice ccapi inout for a nonworking call?

//Suresh Please rate all the useful posts.
New Member

Last time I enabled debug

Last time I enabled debug ccsip all the server "crashed"..I can send you a  debug ccsip messages + debug voice ccapi inout though..You can find it in the attachment. The flow of the conversation goes like this:

phone in A (pA1) calls phone in M (pM). pM transfers call to another phone in A (pA2). pM talks to pA2 for a while and hangs up (expecting that the conversation would flow from pA1 to pA2). However, pA2 receives a hang up, and the pM gets a ring back from the first call.

Thanks

 

Hi Gabriel, had a quick look

Hi Gabriel,

 

had a quick look of the file and found no relevant error messages. however from the problem description given in the recent post, it seems to be with softswitch M.

 

when ph-A1 is out on hold by ph-M, did the ph-A1 hear any Tone or Music On Hold?

 

what happened here is when ph-M puts the ph-A1 on hold, CUBE should receive the INVITE message  with "c=0.0.0.0 or a=sendonly " from softswitch-M

but here is no sdp to indicate the softswitch-A to stop the media.

Below is the faulty invite received from M (without c=0.0.0.0 or a=sendonly).

 

Jun 12 14:19:40.874: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:3122@10.0.10.254:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.171:5060;branch=z9hG4bK1080319056-331834488
Max-Forwards: 70
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
From: <sip:7422@10.0.10.171>;tag=0_1072219056-331834482
To: "Gabriel Lema" <sip:3122@10.0.10.254>;tag=E205CFBC-2260
Call-ID: 68787AFD-F17311E3-B23395C0-9B8997B@10.0.10.254
CSeq: 2 INVITE
Min-SE: 1800
Session-Expires: 2000;Refresher=uac
Supported: timer
Contact: <sip:7422@10.0.10.171:5060;transport=udp>
Content-Length: 0

 

 

>> so please crosscheck the configurations of softswitch M side to identify the issue.

 

//Suresh

Please rate all the helpful posts

//Suresh Please rate all the useful posts.
New Member

Yes, ph-A1 hears a hold music

Yes, ph-A1 hears a hold music. 

Thanks for all your help Suresh!

New Member

One more thing..what should I

One more thing..what should I look for in the SDP configuration of Softswitch M?

Thanks

when the softswitch M puts

when the softswitch M puts the call of ph-A1 on hold, softswitch-M will send out the INVITE message to CUBE and the sdp of that message should have c=0.0.0.0 or a=sendonly. This is indication to CUBE that the call was put on hold.

//Suresh Please rate all the useful posts.
New Member

I understand, but how does it

I understand, but how does it know that it should put the call on hold then?I can hear the music, so the call is put on hold.

Because it received another invite while being in a call?

Thanks

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