I have changed firewall policy's with voip profile strict / default / and without voip profile.
If you add a VoIP profile, SIP traffic bypasses the SIP session helper and is processed by the SIP ALG. To use the SIP session helper you must not add a VoIP profile to the security policy. Of course i found things about the session helper. Now we change the DTMF standard in the phone to SIP INFO (RFC-2976) and not the default RFC2833 If I call any other external number, with a IVR then DTMF is working normaly.
Somehow the door-intercom don't received the DTMF code The phones and the intercom are connected to an external sip server.īut to open the door, you have to response the call DTMF with dail 2 times zero. Question 2: What should I change on the Grandstream HT812 to make internal numbers work fine? I've tried to simply changed preferred DTMF method to 'In-audio', but it didn't help.I have a Fortigate 60F and generally voip is working normal. I am experiencing issues with DTMF and my Q-SYS softphone when calling an external voice bridge. DTMF is the signal sent from the IP phone to the network, which is generated when pressing the IP phone's keypad during a call." If DTMF is sent from an IP phone (here Cisco ATA) to the network (here Internet), how could it affect dialing internal numbers by callers, which are sent in the opposite direction-from ATA to ministation? Question 1: According to theory, "DTMF (Dual Tone Multi-frequency), better known as touch-tone, is used for telecommunication signaling over analog telephone lines in the voice-frequency band. He only remembers that he changed DTMF method: he turned off 'RFC2833' and switched to 'Inband' or 'SIP INFO'. But, alas, he doesn't remember what he had done exactly. A sysadmin who previously ran this system had changed some settings on SPA122 on the advice of other people, which solved the problem. First the problem with dialing internal numbers was the same. Prehistory: Earlier instead of Grandstream HT812 was Cisco SPA122. When dialing from IP phones, pressing additional numbers almost always works fine, but when dialing from mobile phones, it almost always fails. In these cases they are switched to the default telephone set (which belongs to a secretary), as if a wrong number was dialed. Problem: When dialing internal numbers, callers are not always switched to the corresponding telephone set. After that he is switched to a telephone set with the corresponding internal number. While an IVR of the ministation is playing, the caller presses some additional internal numbers. Situation: When a caller dials a number which belongs to a hosted PBX, the call lands on the HT812, then it is switched to the ministation. 28 VOIP Troubleshooter LLC, Indepth: Packet Loss Burstiness, 2005.
Internet => Grandstream HT812 with hosted PBX users => analog ministation Panasonic => analog telephone sets RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals (RFC 4733.
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