VOIP Handsets

ArtyLoop

Executive Member
Joined
Dec 18, 2017
Messages
5,447
#61
Hi, just want to follow up this. Do you have any success in setting this up?

I have signed up with freshphone and is porting my Telkom number to them, but didn't know that they don't support multiple devices connecting to the same SIP account at the same time.

So it's either

A. Get a ATA and keep using my existing telephone, OR

B. Setup Freepbx VM to connect to Freshphone and have multiple IP phones connect to the Freepbx VM.

A is definitely more cost effective, but B is more fun... provided that it's possible.
It did work... like a boss...
 

ffeng

Active Member
Joined
Jan 14, 2010
Messages
30
#64
I tried setting up trunk using PJSIP, which did connect successfully to FreshPhone, and I could make and receive call, but it disconnects quickly afterwards.

I then tried setting up trunk using SIP, but failed to have it connect to FreshPhone, below is my current config:

host=sip1.freshphone.co.za
port=5060
username=087xxxxxxx
secret=THE SECRET
type=peer
 

ffeng

Active Member
Joined
Jan 14, 2010
Messages
30
#65
OK just to report on some progress, still some problem though, but I managed to get it to accept incoming call and I can also dial out, however, the call gets disconnected as soon as it's answered :crying:

See below for the trunk settings (CHAN_SIP) I have:

---------------------------------------
Outgoing:
disallow=all
host=sip1.freshphone.co.za
username=2787xxxxxxx
secret=THE SECRET
type=peer
trunk=yes
timezone=Africa/Johannesburg
qualify=yes
context=from-trunk
allow=g729
keepalive=45
prematuremedia=no
progressinband=yes
silencesuppression=no

Incoming:
host=sip1.freshphone.co.za
type=peer
context=from-trunk
qualify=yes
insecure=invite
fromuser=2787xxxxxxx
---------------------------------------

I also had to configure Asterisk Trunk Dial Option to "r" in order to get ring back when calling out.

On the Inbound Routes, I had to select Signal Ringing to YES for the calling party to get ring back when calling me.

Anyone had similar issues setting up freepbx using freshphone? Much appreciated!

Below is the log:

[2019-02-14 18:19:14] VERBOSE[15212][C-00000008] app_dial.c: SIP/freshphone_sip_outgoing-0000000f is ringing
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] app_dial.c: SIP/freshphone_sip_outgoing-0000000f answered SIP/100-0000000e
[2019-02-14 18:19:23] VERBOSE[15236][C-00000008] bridge_channel.c: Channel SIP/freshphone_sip_outgoing-0000000f joined 'simple_bridge' basic-bridge <104844f5-6b12-4776-9b4a-81878fa858c6>
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] bridge_channel.c: Channel SIP/100-0000000e joined 'simple_bridge' basic-bridge <104844f5-6b12-4776-9b4a-81878fa858c6>
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] bridge_channel.c: Channel SIP/100-0000000e left 'simple_bridge' basic-bridge <104844f5-6b12-4776-9b4a-81878fa858c6>
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] app_macro.c: Spawn extension (macro-dialout-trunk, s, 32) exited non-zero on 'SIP/100-0000000e' in macro 'dialout-trunk'
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Spawn extension (from-internal, 082xxxxxxx, 7) exited non-zero on 'SIP/100-0000000e'
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [h@from-internal:1] Macro("SIP/100-0000000e", "hangupcall") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/100-0000000e", "1?theend") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/100-0000000e", "0?Set(CDR(recordingfile)=)") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@macro-hangupcall:4] NoOp("SIP/100-0000000e", "SIP/freshphone_sip_outgoing-0000000f monior file= ") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@macro-hangupcall:5] AGI("SIP/100-0000000e", "attendedtransfer-rec-restart.php,SIP/freshphone_sip_outgoing-0000000f,") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2019-02-14 18:19:23] VERBOSE[15236][C-00000008] bridge_channel.c: Channel SIP/freshphone_sip_outgoing-0000000f left 'simple_bridge' basic-bridge <104844f5-6b12-4776-9b4a-81878fa858c6>
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] res_agi.c: <SIP/100-0000000e>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@macro-hangupcall:6] Hangup("SIP/100-0000000e", "") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/100-0000000e' in macro 'hangupcall'
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-0000000e'
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] app_stack.c: SIP/100-0000000e Internal Gosub(crm-hangup,s,1) start
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@crm-hangup:1] NoOp("SIP/100-0000000e", "Sending Hangup to CRM") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@crm-hangup:2] NoOp("SIP/100-0000000e", "HANGUP CAUSE: 16") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@crm-hangup:3] ExecIf("SIP/100-0000000e", "0?Set(__CRM_VOICEMAIL=)") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@crm-hangup:4] NoOp("SIP/100-0000000e", "MASTER CHANNEL: 1550161151.14 = 1550161151.14") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@crm-hangup:5] GotoIf("SIP/100-0000000e", "0?return") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@crm-hangup:6] Set("SIP/100-0000000e", "__CRM_HANGUP=1") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@crm-hangup:7] AGI("SIP/100-0000000e", "sangomacrm.agi") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] res_agi.c: <SIP/100-0000000e>AGI Script sangomacrm.agi completed, returning 0
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] pbx.c: Executing [s@crm-hangup:8] Return("SIP/100-0000000e", "") in new stack
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-0000000e'
[2019-02-14 18:19:23] VERBOSE[15212][C-00000008] app_stack.c: SIP/100-0000000e Internal Gosub(crm-hangup,s,1)
 

ffeng

Active Member
Joined
Jan 14, 2010
Messages
30
#66
Alright, just to do my final updates.

I have reverted to using PJSIP instead of CHAN SIP, and all is working fine now after resetting the phone (SNOM 370) and re-provisioning the extension.
 
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