[Freeswitch-dev] G723 timer problem
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. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk@hotmail.com Email: shaheryarkh@googlemail.com
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.
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@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 FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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@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@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.
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
-- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk@hotmail.com Email: shaheryarkh@googlemail.com
I'm sorry about my response... I had overlooked that I only did one 30ms implementation in mod_g723_1.c, Anthony added some more to the list so it might actually work correctly. Thanks, Brian 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.
participants (3)
-
Anthony Minessale
-
Brian West
-
Muhammad Shahzad