[Freeswitch-dev] [Freeswitch-users] G723 timer problem

[Muhammad Shahzad]

Great. It works. Thank you so much. Now i can restore my production
deployment plan back to FS.

Thank you.


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.
>> /b
>>
>> On Jul 1, 2009, at 4:04 PM, Muhammad Shahzad wrote:
>>
>> Hi,
>>
>> 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
>> platforms/devices:
>> Linksys/Sipura aka Cisco
>> ShoreTel
>> Sonus/L3
>> 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.
>>
>>
>>
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>>
>
>
> --
> Anthony Minessale II
>
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-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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