[asterisk-bugs] [JIRA] (ASTERISK-13145) [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML

[Florian Hanig (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=235501#comment-235501 ] 

Florian Hanig edited comment on ASTERISK-13145 at 3/1/17 10:13 AM:
-------------------------------------------------------------------

Okay I've tried this.
I've added this in the chan_sip.c
{code}
 /* Start building generic SDP headers */
                ast_str_append(&a_audio, 0, "b=AS:%d\r\n", p->maxcallbitrate);
                if (ast_test_flag(&p->flags[2], SIP_PAGE3_RELAY_NEAREND)) {
                        ast_str_append(&a_audio, 0, "a=label:X-relay-nearend\r\n");
                } else if (ast_test_flag(&p->flags[2], SIP_PAGE3_RELAY_FAREND)) {
                        ast_str_append(&a_audio, 0, "a=label:X-relay-farend\r\n");
                }
{code}

And then I have compiled it again.

But my Problem with the 8821 persists...

I've done and sip debug again after this modification, and the 8821 has again the Session-ID and the "b=AS:4064"...


was (Author: geforce28):
Okay I've tried this.
I've added this in the chan_sip.c
[code]
 /* Start building generic SDP headers */
                ast_str_append(&a_audio, 0, "b=AS:%d\r\n", p->maxcallbitrate);
                if (ast_test_flag(&p->flags[2], SIP_PAGE3_RELAY_NEAREND)) {
                        ast_str_append(&a_audio, 0, "a=label:X-relay-nearend\r\n");
                } else if (ast_test_flag(&p->flags[2], SIP_PAGE3_RELAY_FAREND)) {
                        ast_str_append(&a_audio, 0, "a=label:X-relay-farend\r\n");
                }
[/code]

And then I have compiled it again.

But my Problem with the 8821 persists...

I've done and sip debug again after this modification, and the 8821 has again the Session-ID and the "b=AS:4064"...

> [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML
> -------------------------------------------------------------------------------
>
>                 Key: ASTERISK-13145
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-13145
>             Project: Asterisk
>          Issue Type: New Feature
>          Components: Channels/chan_sip/NewFeature
>            Reporter: David McNett
>            Assignee: Gareth Palmer
>         Attachments: 00_READ_ME_FIRST.txt, cisco-usecallmanager-11.17.0.patch, cisco-usecallmanager-11.17.1.patch, cisco-usecallmanager-11.18.0.patch, cisco-usecallmanager-11.19.0.patch, cisco-usecallmanager-11.20.0.patch, cisco-usecallmanager-11.21.2.patch, cisco-usecallmanager-11.22.0.patch, cisco-usecallmanager-11.23.0.patch, cisco-usecallmanager-11.24.1.patch, cisco-usecallmanager-11.25.0.patch, cisco-usecallmanager-13.10.0.patch, cisco-usecallmanager-13.12.1.patch, cisco-usecallmanager-13.13.0.patch, dialtemplate.xml, featurepolicy.xml, SEP000000000000.cnf.xml, softkeys.xml
>
>
> This patch provides support for Cisco 6900, 7900, 8800 and 9900 series phones using the SIP firmware.
> Available features are: Busy Lamp Field, Off Hook Notification, Call Forward, Do Not Disturb, Huntgroup Login, Call Park (Notify and Monitor), Server-Side Ad-Hoc Conference, Conference List, Kick and Mute/Unmute, Multi-Admin Conference, Multiple Lines via Bulk Register, Immediate Divert, Call Recording, Restart or Reset via CLI, Call Pickup Notification, Call Back, Join Calls, Mallicious Call ID, Quality Reporting Tool and Fail-over/Fail-back.
> Also included is Application Server Events used by non-USECALLMANAGER phones (Call Forward and Do Not Disturb only).
> *Important:* Read the documentation at [http://usecallmanager.nz] to see the additional configuration options required for the phones to operate correctly.



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