Re: [OSL | CCIE_Voice] SIP Timers fine tuning
Hi Robert, Thank you for your reply. In the CUBE Level there is an early offer forced but in the CUCM Level in the Trunk config , I didn't check MTP Required? Will that fix the issue if I checked MTP required and I will use the soft MTP resource then? Thanks, Hesham On 16 June 2013 20:52, Robert Thomas tho...@gmail.com wrote: You should look into Early offer and Early media. Perhaps you might need PRACK enabled, to cut throught the audio before the call connects. Usually your Telco can give the specific requirements you need. On Sun, Jun 16, 2013 at 8:53 AM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear All, I have a SIP Circuit to Verizon and when I call out I hear 3 rings first before the call is actually routed to the PSTN. Also , I have Automated Attendant and when I dial in to the AA the first 3 seconds are cut from the prompt? Any Ideas what parameters should I change to fix that. Thanks, Hesham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Robert Thomas Zamora tho...@gmail.com +50689389544 http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8 CCNP, CCNP Voice ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] SIP Timers fine tuning
Dear All, I have a SIP Circuit to Verizon and when I call out I hear 3 rings first before the call is actually routed to the PSTN. Also , I have Automated Attendant and when I dial in to the AA the first 3 seconds are cut from the prompt? Any Ideas what parameters should I change to fix that. Thanks, Hesham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SIP Timers fine tuning
You should look into Early offer and Early media. Perhaps you might need PRACK enabled, to cut throught the audio before the call connects. Usually your Telco can give the specific requirements you need. On Sun, Jun 16, 2013 at 8:53 AM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear All, I have a SIP Circuit to Verizon and when I call out I hear 3 rings first before the call is actually routed to the PSTN. Also , I have Automated Attendant and when I dial in to the AA the first 3 seconds are cut from the prompt? Any Ideas what parameters should I change to fix that. Thanks, Hesham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Robert Thomas Zamora tho...@gmail.com +50689389544 http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8 CCNP, CCNP Voice ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
