Ring multiple channels simultaneously: The call is handled on a “first come, first served” basis; the first extension to pick up answers the call, and all the other extensions stop ringing and become
available: exten => s,1,Dial(SIP/2000&SIP/2001&SIP/2303))
I try to call a mobil number with 2 sip simultaneously, when the number i call is answered i don’t have any sound.
When i use: “exten => s,1,Dial(SIP/2000&SIP/2001&SIP/2303)” to make simultaneously dial,I thought only one sip is charged(“first come, first served”),when first sip pick up the call, the secound sip is stop calling. But no, both are charged. It is any mistake in my configuration, or I need to use another command ?
The charging issue will relate to the ITSP’s policies. The PSTN policy equates to starting charging when the far end generates the equivalent of 200 OK. You cannot control that, except in as much as the calls should get cancelled once one answers.
If insecure=very is working, you are using an obsolete version of Asterisk, so knowing the version number may be important. (Also note that most cases of insecure=port,invite (the new form) should be insecure=invite.) This will cause problems with incoming calls.
A detailed explanation of what is wrong will require the SIP and SDP for all the calls.
Finally, this is the wrong forum for support questions.
I tried after to call with one sip a mobil number/landline and i received this massage:
Executing [00448706161166@default:1] Dial(“SIP/3CX-0000007f”, “SIP/811280112/00448706161166”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/811280112/00448706161166
– SIP/811280112-00000080 is making progress passing it to SIP/3CX-0000007f
– SIP/811280112-00000080 answered SIP/3CX-0000007f
– Locally bridging SIP/3CX-0000007f and SIP/811280112-00000080
[Jun 21 17:31:31] WARNING[2816]: chan_sip.c:3351 __sip_xmit: sip_xmit of 0x96fcc10 (len 836) to (null) returned -1: Invalid argument
[Jun 21 17:31:31] WARNING[2816]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 0cb0b8b3121d469b4923440d2db834dd@127.0.0.1:5060 for seqno 102 ( Critical Request)
– See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 31999ms with no response
== Spawn extension (default, 00448706161166, 1) exited non-zero on ‘SIP/3CX-0000007f’
As noted, wrong settings of insecure tend to cause problems for incoming calls, not outgoing ones.
This suggests a bug, but you will need full SIP channel debugging before you can report it. (core set debug 5 chan_sip and sip set debug on)
At a minimum, you need sip history to work out what this is. It is quite likely to be a consequence of the above problem, but could be the result of trying to issue a re-invite to something that has broken re-invite handling (neither handles nor rejects them).