VincentdeWit
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Resource Library: Monster UI Apps for KAZOO
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Posts posted by VincentdeWit
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21 hours ago, amn said:
FusionPBX templates appear to be using Smarty? Did you convert them all to Twig?
Not yet, just a few for testing purposes. I can share some code and upload it to Github, if I find some spare time this week.
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16 hours ago, amn said:
I would be interested in seeing what you are doing on the back end. What templating system are you using? I am looking at maybe using Twig. Using MySQL as the primary database and php-on-couch for tying in with CouchDB.
https://github.com/PHP-on-Couch/PHP-on-Couch
If you are stuck on how to do something have a look at how FusionPBX does it for an example. The code on that project is fairly easy to follow.
Thanks for the tip! I've been using the template files provided by FusionPBX, they've saved a ton of time since they are already in their Github repo.
I've been familiar with Twig, and using the node-twig implementation for templating. It's really easy to understand and one of the simplest out there.If you're looking into a NodeJS API, take a look at Cradle as CouchDB client.
@Tom Hmm, I'll look into that one. The thing I'm more concerned about is security wise for provisioning, such as if I know a phone's MAC I can basically lookup every phone's SIP credentials. I could off-course apply IP address filtering (store an array of IP addresses in the account DB, but sounds too complex (how about different branches, remote working, VPN etc.)).
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Been working on this for a while now, and found out that the existing solutions are a bit outdated.
What we did is build a NodeJS API (based on Express) together with MongoDB where we store information such as MAC address, Account ID and Owner ID (if applicable).
Our beta solution now basically provides a few things;
- Provisioning based on MAC address
- Remote phonebook support linked to users and groups in Kazoo (CouchDB layer)
- Custom editable fields, such as IP address, DNS, Language etcThere still are some things that we're not sure of on how it would work best; such as security, scalability and overall performance.
Would be happy to share code when it's fully up and running.
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Nice, I'll test it later!
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Hmm, thanks, forgot to remove the kazoo-configs-kamailio package. SDP ip's seems to be okay now, but still no audio. Any other clues?
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Thanks for the update! Didn't seem to solve anything yet...
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19 minutes ago, Tom said:
At this point, I'm thinking that the SDP fix doesn't do anything anymore, which doesn't surprise me because I found it on the old forum from 2015. I'm grading programs for a class I'm teaching right now, but I'll try to experiment later today.
Hahaha, good luck with that and thanks so far! Let me know if you need anything.
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Attached the kazoo wireshark trace and the sipinterface.xml
sipinterface.xml
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Hmmm, the infrastructure is indeed hosted on a cloud provider like Google/AWS. All the servers are connected through a virtual router and all have their own public ip (62.212.x.x). Enclosed I've attached a small network diagram, maybe that helps out to understand how it looks. The IP that ifconfig.io/ip returns are the public ip's assigned to the virtual machines.
Also attached the Freeswitch log for 1 call. As far as I can see, the IP noted is the internal freeswitch ip? So that should be the problem? The myippbx.io is btw the realm used (and has an a record to the kamailio/kazoo ip).
Thanks so far Tom!
recv 501 bytes from udp/[172.29.40.239]:5060 at 23:28:13.806402: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Record-Route: <sip:[PUBLIC_IP_KAZOO1/62.212.x.1];lr=on;ftag=tU3FvcKtrytNg> Via: SIP/2.0/UDP [PUBLIC_IP_FS1/62.212.x.3]:11000;received=172.29.40.216;rport=11000;branch=z9hG4bKH5jZ90et7HjSN From: "Vincent de Wit" <sip:2000@myippbx.io>;tag=tU3FvcKtrytNg To: <sip:wouter_t46@myippbx.io>;tag=10-4796B91D-5A06360D000AD09A-03EFE700 Call-ID: d26ad858-c66e-11e7-8e22-8f397b1eade2 CSeq: 114842182 INVITE Server: call.center proxy Contact: <sip:46.19.210.33:6050;transport=udp> Content-Length: 0 ------------------------------------------------------------------------ 2017-11-10 23:28:13.798614 [DEBUG] sofia.c:7084 Channel sofia/sipinterface_1/wouter_t46@myippbx.io entering state [proceeding][180] 2017-11-10 23:28:13.798614 [NOTICE] sofia.c:7192 Ring-Ready sofia/sipinterface_1/wouter_t46@myippbx.io! 2017-11-10 23:28:13.798614 [DEBUG] switch_channel.c:3346 (sofia/sipinterface_1/wouter_t46@myippbx.io) Callstate Change DOWN -> RINGING 2017-11-10 23:28:13.818619 [INFO] switch_ivr_originate.c:1215 Sending early media 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[opus:116:48000:20:0:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[G7221:115:32000:20:48000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[G7221:107:16000:20:32000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[G722:9:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[GSM:3:8000:20:13200:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [GSM:3:8000:20:13200:1] ++++ is saved as a match 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[SPEEX:99:8000:20:24600:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[opus:116:48000:20:0:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[G7221:115:32000:20:48000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[G7221:107:16000:20:32000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[G722:9:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[PCMU:0:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[PCMA:8:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[GSM:3:8000:20:13200:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[SPEEX:99:8000:20:24600:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [SPEEX:99:8000:20:24600:1] ++++ is saved as a match 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:97:8000:30:0:1]/[opus:116:48000:20:0:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:97:8000:30:0:1]/[G7221:115:32000:20:48000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:97:8000:30:0:1]/[G7221:107:16000:20:32000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:97:8000:30:0:1]/[G722:9:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:97:8000:30:0:1]/[PCMU:0:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:97:8000:30:0:1]/[PCMA:8:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:97:8000:30:0:1]/[GSM:3:8000:20:13200:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:97:8000:30:0:1]/[SPEEX:99:8000:20:24600:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G7221:115:32000:20:48000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G7221:107:16000:20:32000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[SPEEX:99:8000:20:24600:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G7221:115:32000:20:48000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G7221:107:16000:20:32000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[SPEEX:99:8000:20:24600:1] 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 101@8000 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:3061 Set Codec sofia/sipinterface_1/vincent_t46@myippbx.io PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-11-10 23:28:13.818619 [DEBUG] switch_core_codec.c:111 sofia/sipinterface_1/vincent_t46@myippbx.io Original read codec set to PCMU:0 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 101@8000 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:4767 sofia/sipinterface_1/vincent_t46@myippbx.io Set 2833 dtmf send payload to 101 recv payload to 101 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:6878 AUDIO RTP [sofia/sipinterface_1/vincent_t46@myippbx.io] 172.29.40.216 port 19976 -> 31.161.159.145 port 4288 codec: 0 ms: 20 2017-11-10 23:28:13.818619 [DEBUG] switch_rtp.c:4111 Starting timer [soft] 160 bytes per 20ms 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:7179 sofia/sipinterface_1/vincent_t46@myippbx.io Set 2833 dtmf send payload to 101 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:7186 sofia/sipinterface_1/vincent_t46@myippbx.io Set 2833 dtmf receive payload to 101 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:7209 sofia/sipinterface_1/vincent_t46@myippbx.io Set rtp dtmf delay to 40 2017-11-10 23:28:13.818619 [NOTICE] sofia_media.c:92 Pre-Answer sofia/sipinterface_1/vincent_t46@myippbx.io! 2017-11-10 23:28:13.818619 [DEBUG] kazoo_event_stream.c:245 increased event stream buffer size to 4747 2017-11-10 23:28:13.818619 [DEBUG] kazoo_event_stream.c:245 increased event stream buffer size to 4747 2017-11-10 23:28:13.818619 [DEBUG] switch_channel.c:3474 (sofia/sipinterface_1/vincent_t46@myippbx.io) Callstate Change RINGING -> EARLY 2017-11-10 23:28:13.818619 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/sipinterface_1/vincent_t46@myippbx.io. 2017-11-10 23:28:13.818619 [DEBUG] mod_sofia.c:2364 Ring SDP: v=0 o=FreeSWITCH 1510336517 1510336518 IN IP4 172.29.40.216 s=FreeSWITCH c=IN IP4 172.29.40.216 t=0 0 m=audio 19976 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2017-11-10 23:28:13.818619 [DEBUG] switch_ivr_originate.c:1273 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2017-11-10 23:28:13.818619 [DEBUG] switch_core_codec.c:223 sofia/sipinterface_1/vincent_t46@myippbx.io Push codec L16:100 2017-11-10 23:28:13.818619 [DEBUG] switch_ivr_originate.c:1342 Play Ringback Tone [%(2000,4000,440,480)] send 1309 bytes to udp/[172.29.40.2]:5060 at 23:28:13.826697: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP [PUBLIC_IP_KAZOO1/62.212.x.1]:5060;branch=z9hG4bK6fc4.12bb88a6faccc04fbbe475c618cce258.0;received=172.29.40.2 Via: SIP/2.0/UDP 31.161.159.145:4262;received=31.161.159.145;branch=z9hG4bK-524287-1---bb3fdd96cae67130;rport=4262 Record-Route: <sip:[PUBLIC_IP_KAZOO1/62.212.x.1];lr=on;ftag=ea86c925> From: "2000"<sip:vincent_t46@myippbx.io;transport=UDP>;tag=ea86c925 To: <sip:2001@myippbx.io;transport=UDP>;tag=Q0Q5pU0F13QXD Call-ID: 4vWX5fJDU_tTUQnl2KQOvQ.. CSeq: 1 INVITE Contact: <sip:2001@[PUBLIC_IP_FS1/62.212.x.3]:11000;transport=udp> User-Agent: 2600hz Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 222 Remote-Party-ID: "2001" <sip:2001@myippbx.io>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1510336517 1510336518 IN IP4 172.29.40.216 s=FreeSWITCH c=IN IP4 172.29.40.216 t=0 0 m=audio 19976 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2017-11-10 23:28:13.838618 [DEBUG] sofia.c:7084 Channel sofia/sipinterface_1/vincent_t46@myippbx.io entering state [early][183] 2017-11-10 23:28:13.898613 [DEBUG] kazoo_node.c:1319 Sent erlang message to ecallmgr@JZKJ-Cloud001-I005.leasewebcloud.com <2.9424.2> recv 366 bytes from udp/[172.29.40.2]:1024 at 23:28:14.778147: ------------------------------------------------------------------------ OPTIONS sip:[PUBLIC_IP_FS1/62.212.x.3]:11000 SIP/2.0 Via: SIP/2.0/UDP 62.212.93.117:5060;branch=z9hG4bKb9bc.e665be93000000000000000000000000.0 To: <sip:[PUBLIC_IP_FS1/62.212.x.3]:11000> From: <sip:sipcheck@JZKJ-Cloud001-I006.leasewebcloud.com>;tag=dbffbdc8cba239bd3478767c2adf1fee-65dc CSeq: 10 OPTIONS Call-ID: 2d1d63523f54bd8c-16476@172.29.40.106 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 751 bytes to udp/[172.29.40.2]:5060 at 23:28:14.778234: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 62.212.93.117:5060;branch=z9hG4bKb9bc.e665be93000000000000000000000000.0;received=172.29.40.2 From: <sip:sipcheck@JZKJ-Cloud001-I006.leasewebcloud.com>;tag=dbffbdc8cba239bd3478767c2adf1fee-65dc To: <sip:[PUBLIC_IP_FS1/62.212.x.3]:11000>;tag=Kvj2FFX1c0yjF Call-ID: 2d1d63523f54bd8c-16476@172.29.40.106 CSeq: 10 OPTIONS Contact: <sip:[PUBLIC_IP_FS1/62.212.x.3]:11000> User-Agent: 2600hz Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 366 bytes from udp/[172.29.40.2]:1024 at 23:28:14.778308: ------------------------------------------------------------------------ OPTIONS sip:[PUBLIC_IP_FS1/62.212.x.3]:11000 SIP/2.0 Via: SIP/2.0/UDP 62.212.93.117:5060;branch=z9hG4bK34ac.66d53965000000000000000000000000.0 To: <sip:[PUBLIC_IP_FS1/62.212.x.3]:11000> From: <sip:sipcheck@JZKJ-Cloud001-I006.leasewebcloud.com>;tag=dbffbdc8cba239bd3478767c2adf1fee-df7d CSeq: 10 OPTIONS Call-ID: 2d1d63523f54bd91-16476@172.29.40.106 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 751 bytes to udp/[172.29.40.2]:5060 at 23:28:14.778347: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 62.212.93.117:5060;branch=z9hG4bK34ac.66d53965000000000000000000000000.0;received=172.29.40.2 From: <sip:sipcheck@JZKJ-Cloud001-I006.leasewebcloud.com>;tag=dbffbdc8cba239bd3478767c2adf1fee-df7d To: <sip:[PUBLIC_IP_FS1/62.212.x.3]:11000>;tag=r9gyrpHKycegS Call-ID: 2d1d63523f54bd91-16476@172.29.40.106 CSeq: 10 OPTIONS Contact: <sip:[PUBLIC_IP_FS1/62.212.x.3]:11000> User-Agent: 2600hz Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 779 bytes from udp/[172.29.40.239]:5060 at 23:28:15.715714: ------------------------------------------------------------------------ SIP/2.0 200 OK Record-Route: <sip:[PUBLIC_IP_KAZOO1/62.212.x.1];lr=on;ftag=tU3FvcKtrytNg> Via: SIP/2.0/UDP [PUBLIC_IP_FS1/62.212.x.3]:11000;received=172.29.40.216;rport=11000;branch=z9hG4bKH5jZ90et7HjSN From: "Vincent de Wit" <sip:2000@myippbx.io>;tag=tU3FvcKtrytNg To: <sip:wouter_t46@myippbx.io>;tag=10-4796B91D-5A06360D000AD09A-03EFE700 Call-ID: d26ad858-c66e-11e7-8e22-8f397b1eade2 CSeq: 114842182 INVITE Server: call.center proxy Contact: <sip:46.19.210.33:6050;transport=udp> Content-Type: application/sdp Content-Length: 250 v=0 o=sems 1485264595 1326795701 IN IP4 46.19.210.33 s=sems c=IN IP4 46.19.210.33 t=0 0 m=audio 17032 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=direction:both ------------------------------------------------------------------------ 2017-11-10 23:28:15.708616 [DEBUG] sofia.c:7084 Channel sofia/sipinterface_1/wouter_t46@myippbx.io entering state [completing][200] 2017-11-10 23:28:15.708616 [DEBUG] sofia.c:7094 Remote SDP: v=0 o=sems 1485264595 1326795701 IN IP4 46.19.210.33 s=sems c=IN IP4 46.19.210.33 t=0 0 m=audio 17032 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:both send 478 bytes to udp/[172.29.40.239]:5060 at 23:28:15.716931: ------------------------------------------------------------------------ ACK sip:46.19.210.33:6050;transport=udp SIP/2.0 Via: SIP/2.0/UDP [PUBLIC_IP_FS1/62.212.x.3]:11000;rport;branch=z9hG4bKjecrBvZX4t8BH Route: <sip:[PUBLIC_IP_KAZOO1/62.212.x.1];lr=on;ftag=tU3FvcKtrytNg> Max-Forwards: 70 From: "Vincent de Wit" <sip:2000@myippbx.io>;tag=tU3FvcKtrytNg To: <sip:wouter_t46@myippbx.io>;tag=10-4796B91D-5A06360D000AD09A-03EFE700 Call-ID: d26ad858-c66e-11e7-8e22-8f397b1eade2 CSeq: 114842182 ACK Contact: <sip:mod_sofia@[PUBLIC_IP_FS1/62.212.x.3]:11000> Content-Length: 0 ------------------------------------------------------------------------ 2017-11-10 23:28:15.708616 [DEBUG] sofia.c:7084 Channel sofia/sipinterface_1/wouter_t46@myippbx.io entering state [ready][200] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G7221:115:32000:20:48000:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G7221:107:16000:20:32000:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G7221:115:32000:20:48000:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G7221:107:16000:20:32000:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 101@8000 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:3061 Set Codec sofia/sipinterface_1/wouter_t46@myippbx.io PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-11-10 23:28:15.708616 [DEBUG] switch_core_codec.c:111 sofia/sipinterface_1/wouter_t46@myippbx.io Original read codec set to PCMU:0 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 101@8000 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:4767 sofia/sipinterface_1/wouter_t46@myippbx.io Set 2833 dtmf send payload to 101 recv payload to 101 2017-11-10 23:28:15.708616 [DEBUG] switch_core_media.c:6878 AUDIO RTP [sofia/sipinterface_1/wouter_t46@myippbx.io] 172.29.40.216 port 32222 -> 46.19.210.33 port 17032 codec: 0 ms: 20 2017-11-10 23:28:15.708616 [DEBUG] switch_rtp.c:4111 Starting timer [soft] 160 bytes per 20ms 2017-11-10 23:28:15.718615 [DEBUG] switch_core_media.c:7179 sofia/sipinterface_1/wouter_t46@myippbx.io Set 2833 dtmf send payload to 101 2017-11-10 23:28:15.718615 [DEBUG] switch_core_media.c:7186 sofia/sipinterface_1/wouter_t46@myippbx.io Set 2833 dtmf receive payload to 101 2017-11-10 23:28:15.718615 [DEBUG] switch_core_media.c:7209 sofia/sipinterface_1/wouter_t46@myippbx.io Set rtp dtmf delay to 40 2017-11-10 23:28:15.718615 [NOTICE] sofia.c:8218 Channel [sofia/sipinterface_1/wouter_t46@myippbx.io] has been answered 2017-11-10 23:28:15.718615 [DEBUG] kazoo_event_stream.c:245 increased event stream buffer size to 4827 2017-11-10 23:28:15.718615 [DEBUG] switch_channel.c:3773 (sofia/sipinterface_1/wouter_t46@myippbx.io) Callstate Change RINGING -> ACTIVE 2017-11-10 23:28:15.718615 [DEBUG] kazoo_event_stream.c:245 increased event stream buffer size to 4827 2017-11-10 23:28:15.718615 [DEBUG] switch_core_codec.c:248 sofia/sipinterface_1/vincent_t46@myippbx.io Restore previous codec PCMU:0. 2017-11-10 23:28:15.718615 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/sipinterface_1/vincent_t46@myippbx.io. 2017-11-10 23:28:15.718615 [DEBUG] mod_sofia.c:850 Local SDP sofia/sipinterface_1/vincent_t46@myippbx.io: v=0 o=FreeSWITCH 1510336517 1510336519 IN IP4 172.29.40.216 s=FreeSWITCH c=IN IP4 172.29.40.216 t=0 0 m=audio 19976 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2017-11-10 23:28:15.718615 [NOTICE] switch_ivr_originate.c:3647 Channel [sofia/sipinterface_1/vincent_t46@myippbx.io] has been answered send 1281 bytes to udp/[172.29.40.2]:5060 at 23:28:15.724500: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP [PUBLIC_IP_KAZOO1/62.212.x.1]:5060;branch=z9hG4bK6fc4.12bb88a6faccc04fbbe475c618cce258.0;received=172.29.40.2 Via: SIP/2.0/UDP 31.161.159.145:4262;received=31.161.159.145;branch=z9hG4bK-524287-1---bb3fdd96cae67130;rport=4262 Record-Route: <sip:[PUBLIC_IP_KAZOO1/62.212.x.1];lr=on;ftag=ea86c925> From: "2000"<sip:vincent_t46@myippbx.io;transport=UDP>;tag=ea86c925 To: <sip:2001@myippbx.io;transport=UDP>;tag=Q0Q5pU0F13QXD Call-ID: 4vWX5fJDU_tTUQnl2KQOvQ.. CSeq: 1 INVITE Contact: <sip:2001@[PUBLIC_IP_FS1/62.212.x.3]:11000;transport=udp> User-Agent: 2600hz Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 222 Remote-Party-ID: "Wouter Grentziu" <sip:2001@myippbx.io>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1510336517 1510336518 IN IP4 172.29.40.216 s=FreeSWITCH c=IN IP4 172.29.40.216 t=0 0 m=audio 19976 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2017-11-10 23:28:15.718615 [DEBUG] kazoo_event_stream.c:245 increased event stream buffer size to 5115 2017-11-10 23:28:15.718615 [DEBUG] switch_channel.c:3773 (sofia/sipinterface_1/vincent_t46@myippbx.io) Callstate Change EARLY -> ACTIVE 2017-11-10 23:28:15.718615 [DEBUG] switch_ivr_originate.c:3705 Originate Resulted in Success: [sofia/sipinterface_1/wouter_t46@myippbx.io] 2017-11-10 23:28:15.718615 [DEBUG] kazoo_event_stream.c:245 increased event stream buffer size to 5115 2017-11-10 23:28:15.718615 [DEBUG] sofia.c:7084 Channel sofia/sipinterface_1/vincent_t46@myippbx.io entering state [completed][200] 2017-11-10 23:28:15.718615 [DEBUG] switch_ivr_bridge.c:1614 (sofia/sipinterface_1/wouter_t46@myippbx.io) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2017-11-10 23:28:15.718615 [DEBUG] switch_core_state_machine.c:584 (sofia/sipinterface_1/wouter_t46@myippbx.io) Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 60) 2017-11-10 23:28:15.718615 [DEBUG] switch_core_state_machine.c:653 (sofia/sipinterface_1/wouter_t46@myippbx.io) State EXCHANGE_MEDIA 2017-11-10 23:28:15.718615 [DEBUG] mod_sofia.c:631 SOFIA EXCHANGE_MEDIA 2017-11-10 23:28:15.718615 [DEBUG] kazoo_event_stream.c:245 increased event stream buffer size to 5595 2017-11-10 23:28:15.718615 [DEBUG] kazoo_event_stream.c:245 increased event stream buffer size to 5595 recv 628 bytes from udp/[172.29.40.2]:5060 at 23:28:15.762103: ------------------------------------------------------------------------ ACK sip:2001@[PUBLIC_IP_FS1/62.212.x.3]:11000;transport=udp SIP/2.0 Via: SIP/2.0/UDP [PUBLIC_IP_KAZOO1/62.212.x.1]:5060;branch=z9hG4bK6fc4.d3a23554192a81d6cbf1ec65413ee8d6.0 Via: SIP/2.0/UDP 31.161.159.145:4262;received=31.161.159.145;branch=z9hG4bK-524287-1---956bbbd28d3ac77f;rport=4262 Max-Forwards: 50 Contact: <sip:vincent_t46@31.161.159.145:4262;transport=UDP> To: <sip:2001@myippbx.io;transport=UDP>;tag=Q0Q5pU0F13QXD From: "2000"<sip:vincent_t46@myippbx.io;transport=UDP>;tag=ea86c925 Call-ID: 4vWX5fJDU_tTUQnl2KQOvQ.. CSeq: 1 ACK User-Agent: Zoiper rv2.8.40 Content-Length: 0 X-AUTH-IP: 31.161.159.145 X-AUTH-PORT: 4262 ------------------------------------------------------------------------ 2017-11-10 23:28:15.758614 [DEBUG] sofia.c:7084 Channel sofia/sipinterface_1/vincent_t46@myippbx.io entering state [ready][200]
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Thanks Tom for your reply! That's right, in the meantime I've added the ifconfig (https://ifconfig.io/ip) api in the publicip python and did a reinstall. I now see the public ip address in the config files (both kamailio and freeswitch). Unfortunately it doesn't seem to solve something. The config files are changed and I'm able to call a other extension, but when the connection has been made, there won't be any audio. When the call is forwarded to the voicemail, I'm actually able to hear the greeting.
Any more idea's?
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Hmm, can't do that unfortunately since we run on Leaseweb...
Below I posted my kamailio local.cfg;
################################################################################ ## ROLES ################################################################################ ## Enabled Roles #!trydef DISPATCHER_ROLE #!trydef NAT_TRAVERSAL_ROLE #!trydef REGISTRAR_ROLE #!trydef PRESENCE_ROLE #!trydef RESPONDER_ROLE #!trydef NODES_ROLE #!trydef FAST_PICKUP_ROLE #!trydef PRESENCE_QUERY_ROLE ## Disabled Roles - remove all but the last '#' to enable # # #!trydef TRAFFIC_FILTER_ROLE # # #!trydef WEBSOCKETS_ROLE # # #!trydef TLS_ROLE # # #!trydef ANTIFLOOD_ROLE # # #!trydef RATE_LIMITER_ROLE # # #!trydef ACL_ROLE # # #!trydef MESSAGE_ROLE # # #!trydef PUSHER_ROLE # # #!trydef REGISTRAR_SYNC_ROLE # # #!trydef PRESENCE_SYNC_ROLE # # #!trydef PRESENCE_NOTIFY_SYNC_ROLE ################################################################################ ## SERVER INFORMATION ################################################################################ ## CHANGE "kamailio.2600hz.com" TO YOUR SERVERS HOSTNAME #!substdef "!MY_HOSTNAME!JZKJ-Cloud001-I005.leasewebcloud.com!g" ## CHANGE "127.0.0.1" TO YOUR SERVERS IP ADDRESS ## Usually your public IP. If you need ## to listen on addtional ports or IPs ## add them in "BINDINGS" at the bottom. #!substdef "!MY_IP_ADDRESS!172.29.40.239!g" ## CHANGE "kazoo://guest:guest@127.0.0.1:5672" TO THE AMQP URL ## This should be the primary RabbitMQ server ## in the zone that this server will service. #!substdef "!MY_AMQP_URL!kazoo://rabbitmq:[PASSWORD_REMOVED]@JZKJ-Cloud001-I005.leasewebcloud.com:5672!g" #!substdef "!MY_AMQP_SECONDARY_URL!kazoo://rabbitmq:[PASSWORD_REMOVED]@JZKJ-Cloud001-I006.leasewebcloud.com:5672!g" ## This parameter is only required if you are using websockets ## This value must be present in the HTTP ## Origin header on a new websocket request ## or it will be rejected. If you remove ## it completely the validation will be disabled. #!substdef "!MY_WEBSOCKET_DOMAIN!devnull.lan!g" ################################################################################ ## DISPATCHER FORMAT INFORMATION ################################################################################ ## dispatcher list changed the default storage type from list to table ## this will allow the management of dispatched servers in the DB ## ## if you're using dbtext and can't start kamailio because of dispatcher ## check the format of /etc/kazoo/kamailio/dbtext/dispatcher ## ## # setid(integer) destination(sip uri) flags (integer, optional) ## 1 sip:192.168.16.131:11000 0 ## should converted to ## id(int,auto) setid(int) destination(string) flags(int) priority(int) attrs(string) description(string) ## 1:1:sip\:192.168.16.131\:11000:0:1: : (there is a space at end) ## ## uncommenting this param will allow to continue using dispatcher list # # #!trydef KZ_USE_DISPATCHER_LIST ################################################################################ ## UDP PARAMETERS ################################################################################ ## This parameter is OPTIONAL ## If large UDP packets are dropped by the ## interface try uncommenting this option. ## However, you MUST match this to your ## network adapter! If they do not match, ## all UDP packets over this limit WILL FAIL! ## E.g.: Add MTU=1472 to the /etc/sysconfig/network-scripts/XXX # udp4_raw_mtu = 1472 ################################################################################ ## BINDINGS ################################################################################ ## This parameter is OPTIONAL. ## It will try to locate outbound interface ## on multihomed host. By default forward ## requests use the incoming socket disregarding ## the destination location. When enabled Kamailio ## will select a socket that can reach the ## destination. This reduces performance. mhomed=0 ## YOU SHOULD NOT HAVE TO CHANGE THESE! ## By setting MY_IP_ADDRESS above these will resolve ## to the proper bindings. These are here ## for those with complex layouts who know ## what they are doing :) #!substdef "!UDP_SIP!udp:MY_IP_ADDRESS:5060!g" #!substdef "!TCP_SIP!tcp:MY_IP_ADDRESS:5060!g" #!substdef "!TLS_SIP!tls:MY_IP_ADDRESS:5061!g" #!substdef "!UDP_ALG_SIP!udp:MY_IP_ADDRESS:7000!g" #!substdef "!TCP_ALG_SIP!tcp:MY_IP_ADDRESS:7000!g" #!substdef "!TLS_ALG_SIP!tls:MY_IP_ADDRESS:7001!g" #!substdef "!TCP_WS!tcp:MY_IP_ADDRESS:5064!g" #!substdef "!UDP_WS_SIP!udp:MY_IP_ADDRESS:5064!g" #!substdef "!TLS_WSS!tls:MY_IP_ADDRESS:5065!g" #!substdef "!UDP_WSS_SIP!udp:MY_IP_ADDRESS:5065!g" ## YOU SHOULD NOT HAVE TO CHANGE THESE! ## This will bind the default SIP listeners ## as determined above. The tls-role and ## websocket-role will use the appropriate ## definitions if enabled. These are here ## for those with complex layouts who know ## what they are doing :) listen=UDP_SIP advertise 172.29.40.239:5060 listen=TCP_SIP advertise 172.29.40.239:5060 listen=UDP_ALG_SIP advertise 172.29.40.239:7000 listen=TCP_ALG_SIP advertise 172.29.40.239:7000Already tried editing the advertise IP to the public ip of the kamailio server, no luck.
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Yep, same issue regarding the no audio. Since we are running on a virtual lan, it doesn't seems to work properly. Any idea?
Only one call for a user
in Product Discussion
Posted
We have been trying to get this working as well. Unfortunatly this doesn't seem to work on a user level, it does on a device level tho. Any thoughts @mc_?