SipToSis
Posted:
Wed Feb 11, 2009 7:03 pm
by kddacraker
Hi all,
I came across this
http://www.mhspot.com/sts/siptosis_skyp ... howto.html
do any body know if skype can configure with vicidail if so how to do that ?
thanks in advance
Posted:
Wed Feb 11, 2009 7:32 pm
by mflorell
This just looks like a SIP -> Skype proxy, and it appears as if you would not be able to get too many concurrent calls out of a setup like this.
Since Asterisk just thinks it's SIP you would just set it up like a normal SIP trunk.
GoAutodial and Siptosis??
Posted:
Fri May 06, 2011 11:15 pm
by rhonuatics
Hi!
Newbee here.. Is is possible to use Siptosis with GoAutodial?
I managed to install skype & siptosis on my GoAutodial Test machine by following the instructions from mhspot
I did make it work as a standalone skype installation and Siptosis was registered successfully. If this would work, i'll try it on a multi-channel setup since only on call can be made at a time.
I added a skype trunk/carrier on the Admin->Carriers and set it as the ONLY active carrier.
Question: What context should i use on the Account Entry?
----- Messages on CLI when i try to call a lead on the agent interface
NOTICE[22229]:chan_local.c:508 local_call: No[May 6 23:31:47] == Parsing '/etc/asterisk/manager.conf': [May 6 23:31:47] Found
[May 6 23:31:47] == Manager 'sendcron' logged on from 127.0.0.1
[May 6 23:31:47] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-4515,2", "8600051|F") in new stack
[May 6 23:31:47] > Channel Local/8600051@default-4515,1 was answered.
[May 6 23:31:47] -- Executing [712316523956@default:1] Dial("Local/8600051@default-4515,1", "SIP/skypeuser/12316523956") in new stack
[May 6 23:31:47] -- Called skypeuser/12316523956
[May 6 23:31:47] -- Got SIP response 482 "Loop Detected" back from 192.168.1.241
[May 6 23:31:47] -- Now forwarding Local/8600051@default-4515,1 to 'Local/12316523956@default' (thanks to SIP/skypeuser-00000002)
[May 6 23:31:47] NOTICE[22229]: chan_local.c:508 local_call: No such extension/context 12316523956@default while calling Local channel
[May 6 23:31:47] NOTICE[22229]: app_dial.c:571 wait_for_answer: Failed to dial on local channel for call forward to 'Local'
[May 6 23:31:47] == Everyone is busy/congested at this time (1:0/0/1)
[May 6 23:31:47] == Auto fallthrough, channel 'Local/8600051@default-4515,1' status is 'CHANUNAVAIL'
Thanks in Advance!
-rhon
Posted:
Fri May 06, 2011 11:59 pm
by williamconley
I did make it work as a standalone skype installation and Siptosis was registered successfully.
you'll have to explain what "make it work" means here. did you make a call?
what it boils down to is: Make this work with ASTERISK. Then come ask about GoAutoDial. In essence you just walked into Richard Petty's shop and asked if they do oil changes. (Answer: Yes. Will they change the oil in your BMW: No.)
AFTER you have this working as a functional asterisk trunk, we can certainly help you make it work with Vicidial. (GoAutoDial, by the way, is the installer ... it installs Asterisk and Vicidial just like Vicibox does, then Asterisk makes the calls and Vicidial is the Control software and reason for being here).
Posted:
Sat May 07, 2011 12:37 am
by rhonuatics
what i mean in "make it work" is i made a call using skype installed on the goautodial(vicidial/asterisk) machine to test if i installed it correctly as instructed at mhspot.
unfortunately, i can't make it work on vicidial to use the skype trunk which i created on the admin-> carriers page.
i've checked the asterisk CLI:
Host Username Refresh State Reg.Time
192.168.1.241:5060 skypeuser 105 Registered Sat, 07 May 2011 01:33:21
@striker:
can you please post the procedures? i would like to try it..
Thanks!
pbx in a flash method
Posted:
Sat May 07, 2011 12:44 am
by striker
heres the link i followed
i forget the initial methods , once i remind i will post
http://nerdvittles.com/index.php?p=587
Posted:
Sat May 07, 2011 12:51 am
by rhonuatics
I have a separate test machine with AsteriskNow/FreePBX and siptosis/skype trunk.
This one is working as described at mhspot. i can make outside calls using x-lite and Skype trunk.
Posted:
Sat May 07, 2011 12:56 am
by williamconley
If you DID make it work, post your sip.conf entry and your dial plan from that "working" version (perhaps the command line output from asterisk from the successful call) and we can help you construct your "converted for Vicidial" version.
And when it's all done, you can post a step by step for the next guys (and then there'll be more help for you later, cuz lots of people will use it even if it IS only one channel).
Posted:
Sat May 07, 2011 12:56 am
by rhonuatics
i tried both, from mhspot and nerdvittles.
both methods worked on my AsteriskNow/FreePBX test machine.
what i'm trying to do is to make it also work with Vicidial, which i am seeking help right now.
Thanks!
Posted:
Sat May 07, 2011 1:05 am
by rhonuatics
The sip.conf & extensions.conf files here are auto-generated by FreeBPX. I made most of the configurations using the FreePBX GUI.
Posted:
Sat May 07, 2011 1:17 am
by rhonuatics
here is the CLI output from my Asterisk/FreePBX machine w/ siptosis:
Connected to Asterisk 1.4.35 currently running on sipserver (pid = 2861)
Verbosity is at least 3
-- Registered SIP '1001' at 192.168.1.251 port 40554
-- Executing [76086611376@from-internal:1] Macro("SIP/1001-00000000", "user-callerid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/1001-00000000", "AMPUSER=1001") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/1001-00000000", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/1001-00000000", "1|Set|REALCALLERIDNUM=1001") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/1001-00000000", "AMPUSER=1001") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/1001-00000000", "AMPUSERCIDNAME=Ronnie") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/1001-00000000", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/1001-00000000", "AMPUSERCID=1001") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/1001-00000000", "CALLERID(all)="Ronnie" <1001>") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/1001-00000000", "1?continue") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [s@macro-user-callerid:18] Set("SIP/1001-00000000", "CALLERID(number)=1001") in new stack
-- Executing [s@macro-user-callerid:19] Set("SIP/1001-00000000", "CALLERID(name)=Ronnie") in new stack
-- Executing [s@macro-user-callerid:20] NoOp("SIP/1001-00000000", "Using CallerID "Ronnie" <1001>") in new stack
-- Executing [76086611376@from-internal:2] Set("SIP/1001-00000000", "_NODEST=") in new stack
-- Executing [76086611376@from-internal:3] Macro("SIP/1001-00000000", "record-enable|1001|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/1001-00000000", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/1001-00000000", "0|MacroExit|") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/1001-00000000", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/1001-00000000", "0?IN") in new stack
-- Executing [s@macro-record-enable:16] ExecIf("SIP/1001-00000000", "1|MacroExit|") in new stack
-- Executing [76086611376@from-internal:4] Macro("SIP/1001-00000000", "dialout-trunk|2|6086611376||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/1001-00000000", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1001-00000000", "0?sub-pincheck|s|1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1001-00000000", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/1001-00000000", "DIAL_NUMBER=6086611376") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/1001-00000000", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/1001-00000000", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1001-00000000", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1001-00000000", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/1001-00000000", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/1001-00000000", "outbound-callerid|2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1001-00000000", "0|SetCallerPres|") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1001-00000000", "0|Set|REALCALLERIDNUM=1001") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1001-00000000", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/1001-00000000", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/1001-00000000", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/1001-00000000", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1001-00000000", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1001-00000000", "0|Set|CALLERID(all)=") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/1001-00000000", "0|Set|CALLERID(all)=") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/1001-00000000", "0|Set|CALLERID(all)=") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/1001-00000000", "0|SetCallerPres|prohib_passed_screen") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/1001-00000000", "1|AGI|fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
== fixlocalprefix: Dialpattern . matched. 6086611376 -> 6086611376
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/1001-00000000", "OUTNUM=6086611376") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/1001-00000000", "custom=SIP/skypeuser") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1001-00000000", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/1001-00000000", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1001-00000000", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1001-00000000", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1001-00000000", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/1001-00000000", "SIP/skypeuser/6086611376|300|") in new stack
-- Called skypeuser/6086611376
-- SIP/skypeuser-00000001 answered SIP/1001-00000000
-- Packet2Packet bridging SIP/1001-00000000 and SIP/skypeuser-00000001
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/1001-00000000", "hangupcall|") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/1001-00000000", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/1001-00000000", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/1001-00000000", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/1001-00000000", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-00000000' in macro 'hangupcall'
== Spawn h extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/1001-00000000'
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/1001-00000000' in macro 'dialout-trunk'
== Spawn extension (from-internal, 76086611376, 4) exited non-zero on 'SIP/1001-00000000'
Posted:
Sat May 07, 2011 11:52 pm
by williamconley
And the engine in my car was built by a machine. But that doesn't mean i can drive without it. And if you want us to be able to convert THAT setup to the Vicidial setup, I'm not sure how we can do it without THAT setup.
Find it in the sip.conf and/or list all the entries you made in the FreePBX setup (from which FreePBX generated the sip.conf).