[Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G
Colin Mason
cmason at frontiernetworks.ca
Sat Aug 25 04:13:35 MSD 2012
Here is my issue:
My $${domain} is the IP of my LAN interface (mpls sip_profile) or 192.168.30.40
When trying to pickup a call on a phone registered to the mpls sip_profile it works:
2012-08-24 19:19:39.575310 [ERR] sofia.c:8810 PICK SQL select call_id from sip_dialogs where call_info='appearance-index=1' and ((sip_from_user='B102_1' and sip_from_host='192.168.30.40') or presence_id='B102_1 at 192.168.30.40') and call_id is not null [01e47cbc-68e5-1230-1cbe-b2f5313092f2] [01e47cbc-68e5-1230-1cbe-b2f5313092f2] 1
+--------------------+---------------+---------------+----------------------+--------------------------------------+
| call_info | sip_from_user | sip_from_host | presence_id | call_id |
+--------------------+---------------+---------------+----------------------+--------------------------------------+
| appearance-index=1 | B101_1 | 192.168.30.40 | NULL | NULL |
| appearance-index=1 | B101_1 | 149.x.x.x | B101_1 at 149.x.x.x | 18824d44-9b44e958 at 10.102.44.10 |
| appearance-index=1 | B102_1 | 192.168.30.40 | B102_1 at 192.168.30.40 | 84094a45-68e6-1230-1cbe-b2f5313092f2 |
+--------------------+---------------+---------------+----------------------+--------------------------------------+
When trying to pickup a call on a phone registered to the internet sip_profile it is broken because the invite sip_from_host is the IP address of my internet sip_profile so it finds nothing in the sip_dialog table because the sip_from_host it should be looking for is $${domain}
2012-08-24 17:59:54.175306 [ERR] sofia.c:8810 PICK SQL select call_id from sip_dialogs where call_info='appearance-index=1' and ((sip_from_user='B105_1' and sip_from_host='149.x.x.x') or presence_id='B105_1 at 149.x.x.x') and call_id is not null [(null)] [] 0
+--------------------+---------------+---------------+----------------------+--------------------------------------+
| call_info | sip_from_user | sip_from_host | presence_id | call_id |
+--------------------+---------------+---------------+----------------------+--------------------------------------+
| appearance-index=1 | B105_1 | 192.168.30.40 | B105_1 at 192.168.30.40 | ea837341-68e6-1230-1cbe-b2f5313092f2 |
| appearance-index=1 | B101_1 | 192.168.30.40 | NULL | NULL |
| appearance-index=1 | B101_1 | 149.x.x.x | B101_1 at 149.x.x.x | 99659a3b-d369ebf6 at 10.102.44.10 |
+--------------------+---------------+---------------+----------------------+--------------------------------------+
Any tips?
Colin
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colin Mason
Sent: Friday, August 24, 2012 5:24 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G
Give me some more time and I will get you a full pcap of what is going on.
After more troubleshooting this issue I've found that I can get transferring to work on one of my SIP profiles. I have 2 profiles that phones register to. One is on the internet and one is on my LAN. I have 3 phones registering from my LAN (mpls sip profile) and 3 phones registering from the internet (internet sip profile). There are 6 SIP accounts in total.
if I call the LAN phones (mpls profile) I am able to put a call on hold and pick it up using another phone.
If I call the internet phones (internet profile) I am NOT able to put a call on hold and pick it up using another phone and the behavior I described earlier occurs.
The LAN phones and the internet phones both send an invite when trying to pickup a held call. The only difference in the invites is the domain so this should be a FreeSWITCH configuration issue:
LAN:
INVITE sip:149.x.x.x SIP/2.0
From: "Line 1" <sip:B105_1 at 149.x.x.x>;tag=603fbd37915ee220o0
To: "Line 1" <sip:B105_1 at 149.x.x.x>
Internet:
INVITE sip:192.168.30.40 SIP/2.0
From: "Line 1" <sip:B102_1 at 192.168.30.40>;tag=1c523101af886951o0
To: "Line 1" <sip:B102_1 at 192.168.30.40>
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Don Dawson
Sent: Friday, August 24, 2012 3:40 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G
I am concerned that FS is trying to INVITE the calling user, ours doesn't do that. Is it possible to send us a pcap trace of this?
In the mean time, here is how our SPA phones are configured. We are sharing 3 lines on multiple phones. The following is the same on all phones. If you want, you can configure the 4th button as a private line (which isn't included here). The 3 shared extensions are 121, 122 and 123.
<CTI_Enable>no</CTI_Enable>
<Dialog_SDP_Enable>yes</Dialog_SDP_Enable>
<Linksys_Key_System>no</Linksys_Key_System>
<Share_Call_Appearance_1_>shared</Share_Call_Appearance_1_>
<Share_Call_Appearance_2_>shared</Share_Call_Appearance_2_>
<Share_Call_Appearance_3_>shared</Share_Call_Appearance_3_>
<Extension_1_>1</Extension_1_>
<Extension_2_>2</Extension_2_>
<Extension_3_>3</Extension_3_>
<Short_Name_1_>Line 1</Short_Name_1_>
<Short_Name_2_>Line 2</Short_Name_2_>
<Short_Name_3_>Line 3</Short_Name_3_>
<Share_Ext_1_>shared</Share_Ext_1_>
<Share_Ext_2_>shared</Share_Ext_2_>
<Share_Ext_3_>shared</Share_Ext_3_>
<Shared_User_ID_1_>121</Shared_User_ID_1_>
<Shared_User_ID_2_>122</Shared_User_ID_2_>
<Shared_User_ID_3_>123</Shared_User_ID_3_>
<Display_Name_1_>Line 1</Display_Name_1_>
<Display_Name_2_>Line 2</Display_Name_2_>
<Display_Name_3_>Line 3</Display_Name_3_>
<User_ID_1_>121</User_ID_1_>
<User_ID_2_>122</User_ID_2_>
<User_ID_3_>123</User_ID_3_>
The FS profile
<param name="manage-shared-appearance" value="true"/>
<param name="manage-presence" value="true"/>
<param name="dbname" value="share_presence"/>
<param name="presence-hosts" value="_DISABLED_"/>
<param name="multiple-registrations" value="true"/>
<param name="send-presence-on-register" value="true"/>
On 8/24/2012 12:52 PM, Colin Mason wrote:
Each phone is setup like:
Button 1: user_1
Button 2: user_2
Button 3: user_3
So for the 3 phones with 3 lines each there are 3 SIP accounts in total, not 9. I believe it's setup properly
Colin
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew Cassidy
Sent: Friday, August 24, 2012 1:46 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G
How is the SLA set up?
To my knowledge for 'true' SLA you need to set them all up using the same SIP account. Again, not something I've tried and Don appears to be better equipped to help you than I so may be able to clarify the correct setup for us?
Thanks
On 24 August 2012 18:18, Colin Mason <cmason at frontiernetworks.ca<mailto:cmason at frontiernetworks.ca>> wrote:
Thanks for the replies.
I tried a few of Andrew's suggestions and I was still unable to get call pickup (transfer?) working properly.
Don, I am seeing different behavior. When I press hold on phone 1, I get an invite from phone 1 with SDP of 0.0.0.0 which puts the call on hold.
- Phone 2's button changes to indicate that the call is on hold.
- I then press the button to pickup the call from hold and phone 2 sends an invite to FreeSWITCH with the username associated with that button. (this may be an issue?)
- FreeSWITCH attemps to dial the user associated with the button I pressed on phone 2.
- Because this user already has a call active and I limit all my users to 1 concurrent call, FreeSWITCH rolls over to the second user.
- Phone 1 and phone 2 receive a call on line 2 and line 1 never drops throughout this process.
I suspect my problem is more of a configuration issue on the phone or FreeSWITCH.
Colin
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Don Dawson
Sent: Friday, August 24, 2012 12:20 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G
We have used the shared line feature for some time now. What actually happens when the call is picked up by the second phone sends an INVITE to FS and FS connects the call to the second phone, there isn't a transfer (REFER) between the phones because they are sharing that call.
What we have found is this worked before version 1.2.0 and now there is a problem. Could you verify what you're experiencing when you pick the call on phone 2, is the call drops because FS is sending BYE to the phone that picked up the call and the caller? The other phones have their lights go green because they are sent a NOTIFY with the appearance-state=idle. In the SIP trace you should see an INVITE from the second phone, 200 OK, BYEs and then NOTIFYs. Is so, then you are experiencing the same issue we are.
Mike
On 8/23/2012 9:33 AM, Colin Mason wrote:
So I have been experimenting with a key system for an office with 3 Cisco SPA504G phones and FreeSWITCH 1.2.1. They require 3 shared lines. I have inbound and outbound calls working properly with the shared lines and the phones are configured properly with shared lines, Broadcom etc. Visually everything seems to work with the line notifications on the 3 phones.
My internal profile has:
<param name="manage-shared-appearance" value="true"/>
<param name="manage-presence" value="true"/>
<param name="dbname" value="share_presence"/>
<param name="presence-hosts" value="$${domain}"/>
<param name="multiple-registrations" value="true"/>
<param name="force-register-domain" value="$${domain}"/>
<param name="force-subscription-domain" value="$${domain}"/>
<param name="force-register-db-domain" value="$${domain}"/>
<param name="presence-probe-on-register" value="true"/>
<param name="presence-privacy" value="$${presence_privacy}"/>
<param name="send-presence-on-register" value="true"/>
I am having problems with transferring calls. If I put a call on hold on phone 1 and press the line 1 button on phone 1, the call resud ames just fine. But if I put a call on hold on phone 1 and try to resume the call on phone 2 by pressing the blinking red line 1 button, the phone tries to establish a new call to the user associated with line 1 instead of taking (transferring) the call from phone 1 to phone 2.
I was wondering if anybody could help?
Colin Mason
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_________________________________________________________________________
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