This is a very valid sequence and used with analog lines or T1/E1 CAS lines emulating analog. Because all call progress tones are inband in the audio signal, there is no clear signalling of what is going on with the call. The easy way to deal with this is for the gateway to accept the call immediately and connect the audio to the caller. The caller will then hear the progress of the call.
Some gateways have call progress tone detection and convert to SIP provisional or final responses. In addition, they can detect voice activity which would indicate that the call has been answered. But these detection are not 100% and may lead wrong interpretation (for example not recognizing international call progress tones). About the only advantage I see to having a gateway do call progress detection is that Call Detail Records are more reflective of the reality because when you have the gateway connecting the call every time, the CDR will show the outgoing call as always successful when the caller may not have actually gotten through. Dan ____________________________________________________ Dan Mongrain, eng. Senior Systems Engineer, Public Safety FREQUENTIS USA Inc. 9017 Red Branch Road, Suite 102 Columbia Maryland 21045 Phone +1-301-657-8001 Mobile +1-819-744-0491 Fax +1-301-657-8002 Web www.frequentis.com/usa E-Mail dan.mongr...@frequentis.com Incorporated in the State of Maryland EIN: 52-2178926 CAGE Code: 1XKR9 ____________________________________________________ Confidentiality Notice: This email message, including any attachments, is for the sole use of the intended recipient(s) and contains confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message and associated attachments. -----Original Message----- From: Jing Jiang [mailto:jing.ji...@biamp.com] Sent: October-22-13 7:44 PM To: sip-implementors@lists.cs.columbia.edu Subject: [Sip-implementors] call progress response 180 or 183 We ran into the issue a few times. We sent out "invite" message to a PSTN number. However, we didn't receive 180 or 183, but 200 ok. We know the media gateway would like to start the media immediately for transmitting call progress tone. Is it a valid call sequence? Thanks, Jing ----------------------------------------------------------------------------------- BIAMP SYSTEMS EMAIL NOTICE The information contained in this email and any attachments is confidential and may be subject to copyright or other intellectual property protection. If you are not the intended recipient, you are not authorized to use or disclose this information, and we request that you notify us by reply mail or telephone and delete the original message from your mail system. ----------------------------------------------------------------------------------- _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
