Dear Eric I Got your reqm. but im not able to understood why ur using AGI scripting to place a call or even to do conferencing in Asterisk using SIP. R all your users accessing same Asterisk Server ?? We don;t req PRI if all your users r using Public IP and can access Server. U can directly place a conference using Asterisk's Inbuilt features.
> Hi Mayank, > > Yes, I am trying to conference 2 users in through SIP. I am not using > any Digium card and calls come in from the carrier via SIP. > > First caller would call in and be placed on hold. And I have the unique > name of the channel saved in the database. And the subscriber will get a > text message indicating that you have a call. And if the subscriber > wants to talk to the original caller who is still on hold, he/she will > call into the system and the system would bridge both calls together. I > am getting sporadic results with the bridging. And while the original > caller is on hold , music on hold will not play most of the time. > > I have read that this is a pretty simple feature to do if we use a PRI. > > thanks > > Eric Lee > > > Mayank Mathur wrote: >> hi >> ru looking to do Conferencing b/w users thru SIP / just want 2 >> simultaneous users to get connected thru SIP ?? >> And what Prob ru facing ?? >> Let me know whether if i can help u out . >> >> >> >>> Hi there, >>> >>> I'm trying to bridge 2 SIP channels together via AGI script. The first >>> caller would call in and be placed on hold and the second caller would >>> call in and both the calls gets connected together. >>> >>> But I am having problem with the second caller finding the first >>> channel. >>> >>> Can someone point me to the right direction? >>> >>> thanks >>> >>> Eric >>> >>> _______________________________________________ >>> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >>> >>> asterisk-dev mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-dev >>> >>> >> >> >> > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > -- Regards, Mayank Mathur _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
