1

I'm trying to setup Twilio Elastic SIP Trunking on my Asterisk/ Freepbx instance and have struggle setting up a reliable origination (termination works perfectly fine).

Background - I've done all settings as per twilio guides, tons of forum posts on freepbx and stackoverflow and for the most part it is a working setup.

  1. Outbound from extension/ pbx - Works very well and great clarity.
  2. Inbound from PSTN to Twilio Number: I hear a few short rings first, then few long rings, and it 'may' eventually call my extension. 25% of time the call comes to my extension (and successfully and randomly from one of the Twilio whitelisted IPs, not necessarily the same IP). 75% of the time the call gets disconnected with beeps, call cannot be connected, etc. messages.
  3. When I use the 'Make a Test Call' button to call from Twilio 'Origination' settings page, the call is instant and works perfectly fine.

In #2 above, I notice a 'Failed' log in Twilio for every ring I hear. I cannot decipher much info from the logs (doesn't have any 401, 403, etc). Could anyone help? Why would #3 always work (Twilio's test call button)?

Here is the Twilio log of a failed call:

INVITE sip:+1TWILIONUMBER@MYASTERSIKIP SIP/2.0
Record-Route: <sip:54.172.60.0:5060;lr;ftag=90304243_6772d868_40eb6ad8-c345-48cf-955c-425fd0246d38>
From: <sip:+1MYPSTNPHONE@MYTWILIOORGID.pstn.twilio.com;isup-oli=0;pstn-params=9084818088;cpc=ordinary>;tag=90304243_6772d868_40eb6ad8-c345-48cf-955c-425fd0246d38
To: <sip:+1TWILIONUMBER@MYASTERSIKIP;user=phone>
CSeq: 9141 INVITE
Max-Forwards: 63
Accept: application/sdp
Accept: application/isup
Accept: application/dtmf
Accept: application/dtmf-relay
Accept: multipart/mixed
Session-Expires: 1800
Min-SE: 90
Content-Disposition: session;handling=required
Diversion: <sip:+1TWILIONUMBER@public-vip.us1.twilio.com>;reason=unconditional
Call-ID: 3612263d48dd307c857c2d3c1815ba73@0.0.0.0
Via: SIP/2.0/UDP 54.172.60.0:5060;branch=z9hG4bK18.e94819e6.0
Via: SIP/2.0/UDP 172.18.12.93:5060;rport=5060;received=172.18.12.93;branch=z9hG4bK40eb6ad8-c345-48cf-955c-425fd0246d38_6772d868_285296189381176
Contact: <sip:+1MYPSTNPHONE@172.18.12.93:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
User-Agent: Twilio Gateway
X-Twilio-AccountSid: AC9172c558ab99243b3cccdce67dadd1b9
X-Twilio-ApiVersion: 2010-04-01
Content-Type: application/sdp
X-Twilio-CallSid: CAe5541067a3270dbe765ee9c0b839cec5
Content-Length: 231

v=0
o=- 869153823 869153823 IN IP4 54.172.60.79
s=Twilio Media Gateway
c=IN IP4 54.172.60.79
t=0 0
m=audio 13460 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
INVITE sip:+1TWILIONUMBER@MYASTERSIKIP SIP/2.0
Record-Route: <sip:54.172.60.0:5060;lr;ftag=90304243_6772d868_40eb6ad8-c345-48cf-955c-425fd0246d38>
From: <sip:+1MYPSTNPHONE@MYTWILIOORGID.pstn.twilio.com;isup-oli=0;pstn-params=9084818088;cpc=ordinary>;tag=90304243_6772d868_40eb6ad8-c345-48cf-955c-425fd0246d38
To: <sip:+1TWILIONUMBER@MYASTERSIKIP;user=phone>
CSeq: 9141 INVITE
Max-Forwards: 63
Accept: application/sdp
Accept: application/isup
Accept: application/dtmf
Accept: application/dtmf-relay
Accept: multipart/mixed
Session-Expires: 1800
Min-SE: 90
Content-Disposition: session;handling=required
Diversion: <sip:+1TWILIONUMBER@public-vip.us1.twilio.com>;reason=unconditional
Call-ID: 3612263d48dd307c857c2d3c1815ba73@0.0.0.0
Via: SIP/2.0/UDP 54.172.60.0:5060;branch=z9hG4bK18.e94819e6.0
Via: SIP/2.0/UDP 172.18.12.93:5060;rport=5060;received=172.18.12.93;branch=z9hG4bK40eb6ad8-c345-48cf-955c-425fd0246d38_6772d868_285296189381176
Contact: <sip:+1MYPSTNPHONE@172.18.12.93:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
User-Agent: Twilio Gateway
X-Twilio-AccountSid: AC9172c558ab99243b3cccdce67dadd1b9
X-Twilio-ApiVersion: 2010-04-01
Content-Type: application/sdp
X-Twilio-CallSid: CAe5541067a3270dbe765ee9c0b839cec5
Content-Length: 231

v=0
o=- 869153823 869153823 IN IP4 54.172.60.79
s=Twilio Media Gateway
c=IN IP4 54.172.60.79
t=0 0
m=audio 13460 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
INVITE sip:+1TWILIONUMBER@MYASTERSIKIP SIP/2.0
Record-Route: <sip:54.172.60.0:5060;lr;ftag=90304243_6772d868_40eb6ad8-c345-48cf-955c-425fd0246d38>
From: <sip:+1MYPSTNPHONE@MYTWILIOORGID.pstn.twilio.com;isup-oli=0;pstn-params=9084818088;cpc=ordinary>;tag=90304243_6772d868_40eb6ad8-c345-48cf-955c-425fd0246d38
To: <sip:+1TWILIONUMBER@MYASTERSIKIP;user=phone>
CSeq: 9141 INVITE
Max-Forwards: 63
Accept: application/sdp
Accept: application/isup
Accept: application/dtmf
Accept: application/dtmf-relay
Accept: multipart/mixed
Session-Expires: 1800
Min-SE: 90
Content-Disposition: session;handling=required
Diversion: <sip:+1TWILIONUMBER@public-vip.us1.twilio.com>;reason=unconditional
Call-ID: 3612263d48dd307c857c2d3c1815ba73@0.0.0.0
Via: SIP/2.0/UDP 54.172.60.0:5060;branch=z9hG4bK18.e94819e6.0
Via: SIP/2.0/UDP 172.18.12.93:5060;rport=5060;received=172.18.12.93;branch=z9hG4bK40eb6ad8-c345-48cf-955c-425fd0246d38_6772d868_285296189381176
Contact: <sip:+1MYPSTNPHONE@172.18.12.93:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
User-Agent: Twilio Gateway
X-Twilio-AccountSid: AC9172c558ab99243b3cccdce67dadd1b9
X-Twilio-ApiVersion: 2010-04-01
Content-Type: application/sdp
X-Twilio-CallSid: CAe5541067a3270dbe765ee9c0b839cec5
Content-Length: 231

v=0
o=- 869153823 869153823 IN IP4 54.172.60.79
s=Twilio Media Gateway
c=IN IP4 54.172.60.79
t=0 0
m=audio 13460 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
INVITE sip:+1TWILIONUMBER@MYASTERSIKIP SIP/2.0
Record-Route: <sip:54.172.60.0:5060;lr;ftag=90304243_6772d868_40eb6ad8-c345-48cf-955c-425fd0246d38>
From: <sip:+1MYPSTNPHONE@MYTWILIOORGID.pstn.twilio.com;isup-oli=0;pstn-params=9084818088;cpc=ordinary>;tag=90304243_6772d868_40eb6ad8-c345-48cf-955c-425fd0246d38
To: <sip:+1TWILIONUMBER@MYASTERSIKIP;user=phone>
CSeq: 9141 INVITE
Max-Forwards: 63
Accept: application/sdp
Accept: application/isup
Accept: application/dtmf
Accept: application/dtmf-relay
Accept: multipart/mixed
Session-Expires: 1800
Min-SE: 90
Content-Disposition: session;handling=required
Diversion: <sip:+1TWILIONUMBER@public-vip.us1.twilio.com>;reason=unconditional
Call-ID: 3612263d48dd307c857c2d3c1815ba73@0.0.0.0
Via: SIP/2.0/UDP 54.172.60.0:5060;branch=z9hG4bK18.e94819e6.0
Via: SIP/2.0/UDP 172.18.12.93:5060;rport=5060;received=172.18.12.93;branch=z9hG4bK40eb6ad8-c345-48cf-955c-425fd0246d38_6772d868_285296189381176
Contact: <sip:+1MYPSTNPHONE@172.18.12.93:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
User-Agent: Twilio Gateway
X-Twilio-AccountSid: AC9172c558ab99243b3cccdce67dadd1b9
X-Twilio-ApiVersion: 2010-04-01
Content-Type: application/sdp
X-Twilio-CallSid: CAe5541067a3270dbe765ee9c0b839cec5
Content-Length: 231

v=0
o=- 869153823 869153823 IN IP4 54.172.60.79
s=Twilio Media Gateway
c=IN IP4 54.172.60.79
t=0 0
m=audio 13460 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

Twilio Log View

Twilio log of a successful call:

INVITE sip:+1TWILIONUMBER@MYASTERSIKIP SIP/2.0
Record-Route: <sip:54.172.60.1:5060;lr;ftag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda>
From: <sip:+1MYPSTNPHONE@MYTWILIOORGID.pstn.twilio.com;pstn-params=808181808882;cpc=ordinary>;tag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda
To: <sip:+1TWILIONUMBER@MYASTERSIKIP;user=phone>
CSeq: 31801 INVITE
Max-Forwards: 11
Accept: application/sdp
Accept: application/isup
Accept: application/dtmf
Accept: application/dtmf-relay
Accept: multipart/mixed
Session-Expires: 1800
Min-SE: 90
Content-Disposition: session;handling=required
Diversion: <sip:+1TWILIONUMBER@public-vip.us1.twilio.com>;reason=unconditional
Call-ID: c4d7a4854a1246e4a19d1c56d3f0e169@0.0.0.0
Via: SIP/2.0/UDP 54.172.60.1:5060;branch=z9hG4bKb67c.e698b2c7.0
Via: SIP/2.0/UDP 172.18.3.99:5060;rport=5060;received=172.18.3.99;branch=z9hG4bK4b230251-675c-48ca-92d5-24d1f90c4cda_6772d868_285301165960484
Contact: <sip:+1MYPSTNPHONE@172.18.3.99:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
User-Agent: Twilio Gateway
X-Twilio-AccountSid: AC9172c558ab99243b3cccdce67dadd1b9
X-Twilio-ApiVersion: 2010-04-01
Content-Type: application/sdp
X-Twilio-CallSid: CAe6bf2966fc9f8990f508c0d57b7e7dc9
Content-Length: 233

v=0
o=- 377936330 377936330 IN IP4 54.172.60.201
s=Twilio Media Gateway
c=IN IP4 54.172.60.201
t=0 0
m=audio 11960 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 54.172.60.1:5060;branch=z9hG4bKb67c.e698b2c7.0;received=54.172.60.1;rport=5060
Via: SIP/2.0/UDP 172.18.3.99:5060;rport=5060;received=172.18.3.99;branch=z9hG4bK4b230251-675c-48ca-92d5-24d1f90c4cda_6772d868_285301165960484
Record-Route: <sip:54.172.60.1:5060;lr;ftag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda>
From: <sip:+1MYPSTNPHONE@MYTWILIOORGID.pstn.twilio.com;pstn-params=808181808882;cpc=ordinary>;tag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda
To: <sip:+1TWILIONUMBER@MYASTERSIKIP:5060;user=phone>
Call-ID: c4d7a4854a1246e4a19d1c56d3f0e169@0.0.0.0
CSeq: 31801 INVITE
Server: FPBX-12.0.70(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+1TWILIONUMBER@MYASTERSIKIP:5060>
Content-Length: 0

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 54.172.60.1:5060;branch=z9hG4bKb67c.e698b2c7.0;received=54.172.60.1;rport=5060
Via: SIP/2.0/UDP 172.18.3.99:5060;rport=5060;received=172.18.3.99;branch=z9hG4bK4b230251-675c-48ca-92d5-24d1f90c4cda_6772d868_285301165960484
Record-Route: <sip:54.172.60.1:5060;lr;ftag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda>
From: <sip:+1MYPSTNPHONE@MYTWILIOORGID.pstn.twilio.com;pstn-params=808181808882;cpc=ordinary>;tag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda
To: <sip:+1TWILIONUMBER@MYASTERSIKIP:5060;user=phone>;tag=as3a65945e
Call-ID: c4d7a4854a1246e4a19d1c56d3f0e169@0.0.0.0
CSeq: 31801 INVITE
Server: FPBX-12.0.70(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+1TWILIONUMBER@MYASTERSIKIP:5060>
Content-Length: 0

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 54.172.60.1:5060;branch=z9hG4bKb67c.e698b2c7.0;received=54.172.60.1;rport=5060
Via: SIP/2.0/UDP 172.18.3.99:5060;rport=5060;received=172.18.3.99;branch=z9hG4bK4b230251-675c-48ca-92d5-24d1f90c4cda_6772d868_285301165960484
Record-Route: <sip:54.172.60.1:5060;lr;ftag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda>
From: <sip:+1MYPSTNPHONE@MYTWILIOORGID.pstn.twilio.com;pstn-params=808181808882;cpc=ordinary>;tag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda
To: <sip:+1TWILIONUMBER@MYASTERSIKIP:5060;user=phone>;tag=as3a65945e
Call-ID: c4d7a4854a1246e4a19d1c56d3f0e169@0.0.0.0
CSeq: 31801 INVITE
Server: FPBX-12.0.70(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+1TWILIONUMBER@MYASTERSIKIP:5060>
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.1:5060;branch=z9hG4bKb67c.e698b2c7.0;received=54.172.60.1;rport=5060
Via: SIP/2.0/UDP 172.18.3.99:5060;rport=5060;received=172.18.3.99;branch=z9hG4bK4b230251-675c-48ca-92d5-24d1f90c4cda_6772d868_285301165960484
Record-Route: <sip:54.172.60.1:5060;lr;ftag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda>
From: <sip:+1MYPSTNPHONE@MYTWILIOORGID.pstn.twilio.com;pstn-params=808181808882;cpc=ordinary>;tag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda
To: <sip:+1TWILIONUMBER@MYASTERSIKIP:5060;user=phone>;tag=as3a65945e
Call-ID: c4d7a4854a1246e4a19d1c56d3f0e169@0.0.0.0
CSeq: 31801 INVITE
Server: FPBX-12.0.70(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+1TWILIONUMBER@MYASTERSIKIP:5060>
Content-Type: application/sdp
Content-Length: 483

v=0
o=root 2047240699 2047240699 IN IP4 MYASTERSIKIP
s=Asterisk PBX 13.9.1
c=IN IP4 MYASTERSIKIP
t=0 0
m=audio 14374 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:29fbe7313c8689b0102f9d503d9eb401
a=ice-pwd:7eb4637401aca4580b6985857be28247
a=candidate:Hc0a81d9a 1 UDP 2130706431 192.168.29.154 14374 typ host
a=candidate:Hc0a81d9a 2 UDP 2130706430 192.168.29.154 14375 typ host
a=sendrecv
ACK sip:+1TWILIONUMBER@MYASTERSIKIP:5060 SIP/2.0
Call-ID: c4d7a4854a1246e4a19d1c56d3f0e169@0.0.0.0
CSeq: 31801 ACK
From: <sip:+1MYPSTNPHONE@MYTWILIOORGID.pstn.twilio.com;pstn-params=808181808882;cpc=ordinary>;tag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda
To: <sip:+1TWILIONUMBER@MYASTERSIKIP;user=phone>;tag=as3a65945e
Max-Forwards: 69
User-Agent: Twilio
X-Twilio-CallSid: CAe6bf2966fc9f8990f508c0d57b7e7dc9
Via: SIP/2.0/UDP 54.172.60.1:5060;branch=z9hG4bKb67c.e698b2c7.2
Via: SIP/2.0/UDP 172.18.3.99:5060;rport=5060;received=54.172.60.201;branch=z9hG4bK4b230251-675c-48ca-92d5-24d1f90c4cda_6772d868_285306732657503
Content-Length: 0

BYE sip:+1TWILIONUMBER@MYASTERSIKIP:5060 SIP/2.0
CSeq: 31802 BYE
From: <sip:+1MYPSTNPHONE@MYTWILIOORGID.pstn.twilio.com;pstn-params=808181808882;cpc=ordinary>;tag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda
To: <sip:+1TWILIONUMBER@MYASTERSIKIP;user=phone>;tag=as3a65945e
Call-ID: c4d7a4854a1246e4a19d1c56d3f0e169@0.0.0.0
Max-Forwards: 68
Via: SIP/2.0/UDP 54.172.60.1:5060;branch=z9hG4bK867c.164aadb4.0
Via: SIP/2.0/UDP 172.18.3.99:5060;rport=5060;received=54.172.60.201;branch=z9hG4bK4b230251-675c-48ca-92d5-24d1f90c4cda_6772d868_285317504904769
User-Agent: Twilio Gateway
X-Twilio-CallSid: CAe6bf2966fc9f8990f508c0d57b7e7dc9
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.1:5060;branch=z9hG4bK867c.164aadb4.0;received=54.172.60.1;rport=5060
Via: SIP/2.0/UDP 172.18.3.99:5060;rport=5060;received=54.172.60.201;branch=z9hG4bK4b230251-675c-48ca-92d5-24d1f90c4cda_6772d868_285317504904769
From: <sip:+1MYPSTNPHONE@MYTWILIOORGID.pstn.twilio.com;pstn-params=808181808882;cpc=ordinary>;tag=65942748_6772d868_4b230251-675c-48ca-92d5-24d1f90c4cda
To: <sip:+1TWILIONUMBER@MYASTERSIKIP:5060;user=phone>;tag=as3a65945e
Call-ID: c4d7a4854a1246e4a19d1c56d3f0e169@0.0.0.0
CSeq: 31802 BYE
Server: FPBX-12.0.70(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
tipsytopsy
  • 92
  • 8
  • The first line of your sip method is not readable(for example `=ÊzWÅí ×Ã\U$ E qºj )gHC¿¶^6¬< ÄÄ]Ô4SIP/2.0 100 Trying`), please update logs, so I can take a look. – os11k Jul 05 '16 at 08:25
  • Debug is oftopic on SO.This question is debug only. – arheops Jul 05 '16 at 15:44
  • Thank you for your reply @os11k . I've updated the post with logs without special characters. Please let me know if you can help. – tipsytopsy Jul 05 '16 at 17:28
  • From where you get this logs? From Asterisk server? In logs you can see that twillio tries to send call to Asterisk, but somehow asterisk do not reply to INVITEs, so you are missing something on FreePBX side. I just checked twillio manual and there is stated that you need to create 3-4 trunks configuration for each IP, depending on the region. Do you have this 3-4 separate configs? – os11k Jul 05 '16 at 17:48
  • Logs are from Twilio. Yes, in fact I've created 8 trunks for inbound (even for the oregon IPs). My asterisk IPtables are correctly configured and router NAT has 5060 and 10000-20000 UDP forwarded to my asterisk. I've enabled NAT in all places in FreePBX (nat=force_rport,comedia). No idea how the 'Test call' buton in Twilio instantly rings my phone (they're probably doing a SIP call) while the PSTN to Twilio to Asterisk faces issues. And on Asterisk side, I only see the requests which were successful. No failures. – tipsytopsy Jul 05 '16 at 18:00
  • Ok. To what IP Twillio sends traffic for failed calls and for successful calls? Are you able to see this in your logs? Are this IPs same? Could you please try to capture SIP traffic for failed calls on Asterisk server using ngrep? What router you are using before Asterisk? Do you have access to it? Do you have any possibility to eliminate NAT? – os11k Jul 06 '16 at 06:55
  • Thank you for your reply @os11k. NAT cannot be eliminated since the server is always going to be behind one. I'll enabled some logs tonight and try to remove firewall from the picture tonight. I'll update you with my results. I noticed that UDP in intermittent ()may be due to UDP fragmentation according to Twilio), but TCP does not do through at all. So I think it is an issue on NAT/ Firewall. These logs come for TCP: ul 5 21:05:41 kernel: nf_ct_sip: dropping packetIN= OUT=br0 SRC=54.172.60.1 DST=xx.xx.xx.xx PROTO=TCP – tipsytopsy Jul 06 '16 at 17:32
  • After replacing the firewall and NAT configs, I've the system working partially now. Inbound - Works very well for UDP and TCP. Outbound (Extension to PSTN) - Voice from PBX to PSTN is heard fine. But I cannot hear the PSTN voice on the PBX extension. The only difference I can see between these two scenarios is: Inbound origination - everything is handled using my static IP address and all my firewall rules are working fine. For outbound, my trunk is setup to call myorgId.pstn.twilio.com. Something in NAT (force_rport,comedia) or anything else you have come across before? – tipsytopsy Jul 08 '16 at 13:53
  • @tipsytopsy "With a hostname the phone received the INVITE but always returned 404." https://www.twilio.com/blog/2013/03/build-a-twilio-hard-phone-with-sip-from-twilio-raspberry-pi-asterisk-freepbx-and-the-obihai-obi100.html try specifying an IP address. – Thufir Jul 25 '16 at 10:54

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