Shared Call Appearance and Transferring Calls on Cisco SPA504G
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 resumes 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
I have an SPA303, and the process to transfer (either attended of blind) is to press a key labeled 'xfer' or 'bxfer' then dial the line to transfer to. Aside from that I've not played with SLA on these devices. You can emulate these features using just one line per phone setting up the others as speed dial, blf and call pickup using the syntax fnc=sd+blf+cp;ext=<exten> Make sure you set type to RFC3265_4235 on the Attendant keys tab, and add the relevant dialplan entries to handle the interception of the calls (call pickup prepends ** to the number) Alternatively there's always call parking, too. You could set up 3 slot parking lot, handle the hold event to park the call in a specific slot, then getting the call back again is just a speed dial key. That setup works for me, and is only a suggestion. As I've only got the one device I can't offer any solutions for using SLA at this time. On 23 August 2012 15:33, Colin Mason <cmason@frontiernetworks.ca> 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 resumes 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****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info@cassidywebservices.co.uk> *03300 100 960 *F<info@cassidywebservices.co.uk> *03300 100 961 *E <info@cassidywebservices.co.uk> *andrew@cassidywebservices.co.uk *W <info@cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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 resumes 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Don Dawson Sent: Friday, August 24, 2012 12:20 PM To: freeswitch-users@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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Don Dawson *Sent:* Friday, August 24, 2012 12:20 PM *To:* freeswitch-users@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****
****
_________________________________________________________________________****
Professional FreeSWITCH Consulting Services:****
consulting@freeswitch.org****
http://www.freeswitchsolutions.com****
** **
FreeSWITCH-powered IP PBX: The CudaTel Communication Server****
** **
Official FreeSWITCH Sites****
http://wiki.freeswitch.org****
** **
FreeSWITCH-users mailing list****
FreeSWITCH-users@lists.freeswitch.org****
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users****
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users****
** **
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info@cassidywebservices.co.uk> *03300 100 960 *F<info@cassidywebservices.co.uk> *03300 100 961 *E <info@cassidywebservices.co.uk> *andrew@cassidywebservices.co.uk *W <info@cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@frontiernetworks.ca<mailto:cmason@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@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Don Dawson Sent: Friday, August 24, 2012 12:20 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director [http://c1170247.r47.cf3.rackcdn.com/emailsig.png] T<mailto:info@cassidywebservices.co.uk> 03300 100 960 F<mailto:info@cassidywebservices.co.uk> 03300 100 961 E<mailto:info@cassidywebservices.co.uk> andrew@cassidywebservices.co.uk<mailto:andrew@cassidywebservices.co.uk> W<mailto:info@cassidywebservices.co.uk> www.cassidywebservices.co.uk<http://www.cassidywebservices.co.uk>
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 4^th 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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@frontiernetworks.ca <mailto:cmason@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@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Don Dawson *Sent:* Friday, August 24, 2012 12:20 PM *To:* freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@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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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@149.x.x.x>;tag=603fbd37915ee220o0 To: "Line 1" <sip:B105_1@149.x.x.x> Internet: INVITE sip:192.168.30.40 SIP/2.0 From: "Line 1" <sip:B102_1@192.168.30.40>;tag=1c523101af886951o0 To: "Line 1" <sip:B102_1@192.168.30.40> From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Don Dawson Sent: Friday, August 24, 2012 3:40 PM To: freeswitch-users@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@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@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@frontiernetworks.ca<mailto:cmason@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@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Don Dawson Sent: Friday, August 24, 2012 12:20 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director [http://c1170247.r47.cf3.rackcdn.com/emailsig.png] T<mailto:info@cassidywebservices.co.uk> 03300 100 960 F<mailto:info@cassidywebservices.co.uk> 03300 100 961 E<mailto:info@cassidywebservices.co.uk> andrew@cassidywebservices.co.uk<mailto:andrew@cassidywebservices.co.uk> W<mailto:info@cassidywebservices.co.uk> www.cassidywebservices.co.uk<http://www.cassidywebservices.co.uk> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Here is my issue: My $${domain} is the IP of my LAN interface (mpls sip_profile) or 192.168.30.40 When trying to pickup a call on a phone registered to the mpls sip_profile it works: 2012-08-24 19:19:39.575310 [ERR] sofia.c:8810 PICK SQL select call_id from sip_dialogs where call_info='appearance-index=1' and ((sip_from_user='B102_1' and sip_from_host='192.168.30.40') or presence_id='B102_1@192.168.30.40') and call_id is not null [01e47cbc-68e5-1230-1cbe-b2f5313092f2] [01e47cbc-68e5-1230-1cbe-b2f5313092f2] 1 +--------------------+---------------+---------------+----------------------+--------------------------------------+ | call_info | sip_from_user | sip_from_host | presence_id | call_id | +--------------------+---------------+---------------+----------------------+--------------------------------------+ | appearance-index=1 | B101_1 | 192.168.30.40 | NULL | NULL | | appearance-index=1 | B101_1 | 149.x.x.x | B101_1@149.x.x.x | 18824d44-9b44e958@10.102.44.10 | | appearance-index=1 | B102_1 | 192.168.30.40 | B102_1@192.168.30.40 | 84094a45-68e6-1230-1cbe-b2f5313092f2 | +--------------------+---------------+---------------+----------------------+--------------------------------------+ When trying to pickup a call on a phone registered to the internet sip_profile it is broken because the invite sip_from_host is the IP address of my internet sip_profile so it finds nothing in the sip_dialog table because the sip_from_host it should be looking for is $${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@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@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@149.x.x.x | 99659a3b-d369ebf6@10.102.44.10 | +--------------------+---------------+---------------+----------------------+--------------------------------------+ Any tips? Colin From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@149.x.x.x>;tag=603fbd37915ee220o0 To: "Line 1" <sip:B105_1@149.x.x.x> Internet: INVITE sip:192.168.30.40 SIP/2.0 From: "Line 1" <sip:B102_1@192.168.30.40>;tag=1c523101af886951o0 To: "Line 1" <sip:B102_1@192.168.30.40> From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces@lists.freeswitch.org]> On Behalf Of Don Dawson Sent: Friday, August 24, 2012 3:40 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@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@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@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@frontiernetworks.ca<mailto:cmason@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@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Don Dawson Sent: Friday, August 24, 2012 12:20 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director [http://c1170247.r47.cf3.rackcdn.com/emailsig.png] T<mailto:info@cassidywebservices.co.uk> 03300 100 960 F<mailto:info@cassidywebservices.co.uk> 03300 100 961 E<mailto:info@cassidywebservices.co.uk> andrew@cassidywebservices.co.uk<mailto:andrew@cassidywebservices.co.uk> W<mailto:info@cassidywebservices.co.uk> www.cassidywebservices.co.uk<http://www.cassidywebservices.co.uk> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
A this should be in Jira not ml. B it sounds like Cisco does something evil with SLA so try using the ip of the profile as the domain across the board.... On Aug 24, 2012 7:15 PM, "Colin Mason" <cmason@frontiernetworks.ca> wrote:
Here is my issue:****
** **
My $${domain} is the IP of my LAN interface (mpls sip_profile) or 192.168.30.40****
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When trying to pickup a call on a phone registered to the mpls sip_profile it works:****
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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@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@149.x.x.x | 18824d44-9b44e958@10.102.44.10 |****
| appearance-index=1 | B102_1 | 192.168.30.40 | B102_1@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@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@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@149.x.x.x | 99659a3b-d369ebf6@10.102.44.10 |****
+--------------------+---------------+---------------+----------------------+--------------------------------------+ ****
** **
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Any tips?****
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Colin****
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*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@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. ****
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if I call the LAN phones (mpls profile) I am able to put a call on hold and pick it up using another phone.****
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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@149.x.x.x>;tag=603fbd37915ee220o0****
To: "Line 1" <sip:B105_1@149.x.x.x>****
** **
** **
Internet:****
INVITE sip:192.168.30.40 SIP/2.0****
From: "Line 1" <sip:B102_1@192.168.30.40>;tag=1c523101af886951o0****
To: "Line 1" <sip:B102_1@192.168.30.40>****
** **
** **
** **
** **
** **
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Don Dawson *Sent:* Friday, August 24, 2012 3:40 PM *To:* freeswitch-users@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@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org<freeswitch-users-bounces@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@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@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Don Dawson *Sent:* Friday, August 24, 2012 12:20 PM *To:* freeswitch-users@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****
****
_________________________________________________________________________****
Professional FreeSWITCH Consulting Services:****
consulting@freeswitch.org****
http://www.freeswitchsolutions.com****
****
FreeSWITCH-powered IP PBX: The CudaTel Communication Server****
****
Official FreeSWITCH Sites****
http://wiki.freeswitch.org****
****
FreeSWITCH-users mailing list****
FreeSWITCH-users@lists.freeswitch.org****
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users****
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users****
****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
****
****
-- *Andrew Cassidy BSc (Hons) MBCS SSCA*****
Managing Director****
****
****
*T <info@cassidywebservices.co.uk> *03300 100 960 *F<info@cassidywebservices.co.uk> *03300 100 961****
*E <info@cassidywebservices.co.uk> *andrew@cassidywebservices.co.uk****
*W <info@cassidywebservices.co.uk> *www.cassidywebservices.co.uk****
****
****
_________________________________________________________________________****
Professional FreeSWITCH Consulting Services:****
consulting@freeswitch.org****
http://www.freeswitchsolutions.com****
** **
FreeSWITCH-powered IP PBX: The CudaTel Communication Server****
** **
Official FreeSWITCH Sites****
http://wiki.freeswitch.org****
** **
FreeSWITCH-users mailing list****
FreeSWITCH-users@lists.freeswitch.org****
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users****
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users****
** **
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (4)
-
Andrew Cassidy
-
Anthony Minessale
-
Colin Mason
-
Don Dawson