[Freeswitch-dev] [Freeswitch-users] G723 timer problem
shaheryarkh at googlemail.com
Wed Jul 1 23:36:55 PDT 2009
Great. It works. Thank you so much. Now i can restore my production
deployment plan back to FS.
On Wed, Jul 1, 2009 at 4:30 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:
> try revision 14095 or higher.
> This adds the ability to use g723 at 60ms,
> If this does not work,
> set the param rtp-autofix-timing to true in your sip profile.
> On Wed, Jul 1, 2009 at 4:13 PM, Brian West <brian at freeswitch.org> wrote:
>> You have two choices... set codec neg. to scrooge or get a provider that
>> doesn't lie about the ptime in their SDP.
>> On Jul 1, 2009, at 4:04 PM, Muhammad Shahzad wrote:
>> I am using FS svn revision 14046 and trying to send call from SIP Dialer
>> to a SIP gateway using G723 in passthrough mode. Everything works perfect
>> and destination rings but then call drops with following error on FS CLI,
>> 2009-07-02 02:39:28.790508 [WARNING] mod_sofia.c:807 We were told to use
>> ptime 30 but what they meant to say was 60
>> This issue has so far been identified to happen on the following broken
>> Linksys/Sipura aka Cisco
>> We will try to fix it but some of the devices on this list are so broken
>> who knows what will happen..
>> 2009-07-02 02:39:28.790508 [WARNING] switch_core_codec.c:499 Codec G723
>> Exists but not at the desired implementation. 8000hz 60ms
>> Is there any work around for this or i have downgrade my server back to
>> Asterisk. :'-(
>> Thank you.
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
> Anthony Minessale II
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CISCO Rich Media Communication Specialist (CRMCS)
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Email: shaheryarkh at googlemail.com
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