Hi
Did you ever get this all working? I am in a really similar situation where I can get everything to work in PhonerLite (and Zoiper and MicroSIP), but I can’t work out how to translate those software settings to the chan_sip config inside FreePBX.
I can’t even get the registration to work (which you seem to have achieved).
What does Reports -> Asterisk Info -> Registries show for the trunk?
Do you have any other trunks on your system that are working?
At the Asterisk command prompt, type
sip set debug on
and you should then see the registration attempts and any replies. If it doesn’t make sense, post them here (redact usernames and any other private info).
Have you read this wiki yet? It looks like this has all the info you are looking for.
https://wiki.asterisk.org/wiki/display/DIGIUM/Digium+Phones+and+FreePBX
Best I can see here is:
Yealink sends hold (reINVITE) to SBC via TLS/SRTP
SBC sends request to FreePBX via UDP/RTP
FreePBX sends reply back to SBC via UDP/RTP
SBC sends reply to Yealink via UDP/RTP
Yealink is expecting TLS/SRTP and not getting it and sending a 488 back to the SBC/FreePBX.
Is this issue happening on remote devices not using TLS/SRTP? If the Yealink is set to UDP/RTP only does this issue persist?
Hi. Thanks for helping me out with this… I have a few other regular sip providers registered through PJSIP,
This particular SIP Trunk is the VoIP service for my ISP. Normally this is registered by the CPE mdoem provided by the ISP, but since I have the modem in bridge mode the VoIP functionalities are disabled. I know the registration details and have been able to get them registered and working in PhonerLite, Zoiper and MicroSIP. The ISP seems to use SRV records which seem to work with MicroSIP, so I substituted the proxy domain name with the IP address that PhonerLite was using and that meant I was able to get MicroSIP to register too.
Asterisk info shows “Request sent” for registration, but is never registers.
Outbound
Trunk Name: Telmex
PEER Details: [BLANK]
Inbound
USER Context: +52xxxxxxxxxx
USER Details:
username=+52xxxxxxxxxx
user=+52xxxxxxxxxx
type=friend
secret=############
realm=ims.telmex.com
outboundproxy=189.247.242.147
insecure=invite
host=189.247.242.147
fromuser=+52xxxxxxxxxx
authname=+52xxxxxxxxxx@ims.telmex.com
context=from-sip-external
Register string:
+52xxxxxxxxxx@ims.telmex.com:############:+52xxxxxxxxxx@189.247.242.147
*.*.*.* = my external IP address, not the IP of the FreePBX box (which is 10.0.0.32)
[2018-07-08 09:42:14] NOTICE[1019]: chan_sip.c:15890 sip_reg_timeout: -- Registration for '+52xxxxxxxxxx@189.247.242.147' timed out, trying again (Attempt #50)
Really destroying SIP dialog '72f66926341e700061b302826fed2716@10.0.0.32' Method: REGISTER
Retransmitting #1 (NAT) to 189.247.242.147:5060:
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5160;branch=z9hG4bK0af907e1;rport
Max-Forwards: 70
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=as40f406f2
To: <sip:+52xxxxxxxxxx@ims.telmex.com>
Call-ID: 72f66926341e700061b302826fed2716@10.0.0.32
CSeq: 151 REGISTER
Supported: replaces, timer
User-Agent: FPBX-14.0.3.6(14.7.6)
Expires: 120
Contact: <sip:s@*.*.*.*:5160>
Content-Length: 0
---
Retransmitting #2 (NAT) to 189.247.242.147:5060:
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5160;branch=z9hG4bK0af907e1;rport
Max-Forwards: 70
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=as40f406f2
To: <sip:+52xxxxxxxxxx@ims.telmex.com>
Call-ID: 72f66926341e700061b302826fed2716@10.0.0.32
CSeq: 151 REGISTER
Supported: replaces, timer
User-Agent: FPBX-14.0.3.6(14.7.6)
Expires: 120
Contact: <sip:s@*.*.*.*:5160>
Content-Length: 0
Proxy/Registrar: voipnvcompigl.telmex.net
Domain/Realm: ims.telmex.com
Username: +52xxxxxxxxxx
Password: ############
Authentication name: +52xxxxxxxxxx@ims.telmex.com
SIP Server: ims.telmex.com
SIP Proxy: 189.247.242.147
Username: +52xxxxxxxxxx
Domain: ims.telmex.com
Login: +52xxxxxxxxxx@ims.telmex.com
Password: ############
From here you can see why I ended up using the proxy IP address
I am not sure what the various forbidden and timeout messages are, but PhonerLite does seem to work correctly (I can make and receive calls).
-------------------------------------------
09:52:17,558: R: DNS lookup for 'voipnvcompigl.telmex.net'
start resolving SRV (UDP)...
-------------------------------------------
09:52:17,561: R: DNS lookup for 'slbcompigl.voip.telmex.net'
189.247.242.147:5060 (TTL=1699)
-------------------------------------------
09:52:17,561: R: open UDP port (SIP): 5060
-------------------------------------------
09:52:17,562: R: open TCP port (TLS listen): 5061
-------------------------------------------
09:52:17,562: R: open TCP port (TCP listen): 5060
09:52:17,611: Listen Confirm: 0E 00 08 00 05 81 9E 02 01 00 00 00 00 00
09:52:17,611: Listen Confirm
-------------------------------------------
09:52:17,563: R: open UDP port (mDNS): 5353
09:52:17,621: Facility Confirm: 1A 00 08 00 80 81 A0 02 01 00 00 00 00 00 03 00 09 00 00 06 00 00 3D 01 00 00
09:52:17,621: Facility Confirm (Supplementary Services)
09:52:17,621: Facility Request: 16 00 08 00 80 80 A1 02 01 00 00 00 03 00 07 01 00 04 3D 01 00 00
09:52:17,621: Facility Request (Listen To Supplementary Services)
09:52:17,621: Get Supported Services: success
-------------------------------------------
09:52:17,563: T: 189.247.242.147:5060 (UDP)
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.50:5060;branch=z9hG4bK80a6192d2c81e81187bbbb0be53ecaeb;rport
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=1934202406
To: <sip:+52xxxxxxxxxx@ims.telmex.com>
Call-ID: 80A6192D-2C81-E811-87B8-BB0BE53ECAEB@10.0.0.50
CSeq: 326 REGISTER
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>;+sip.instance="<urn:uuid:00D1773F-C27F-E811-97E2-EED72484AD05>"
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 900
Content-Length: 0
-------------------------------------------
09:52:17,563: T: mDNS refresh: sip:+52xxxxxxxxxx@ims.telmex.com = 169.254.106.102:5060, ttl=900
SIPPER for PhonerLite
09:52:17,621: Facility Confirm: 16 00 08 00 80 81 A1 02 01 00 00 00 00 00 03 00 05 01 00 02 00 00
09:52:17,621: Facility Confirm (Supplementary Services)
09:52:17,621: Listen: success
-------------------------------------------
09:52:17,621: T: 189.247.242.147:5060 (UDP)
SUBSCRIBE sip:+52xxxxxxxxxx@ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.50:5060;branch=z9hG4bK80a6192d2c81e81187bcbb0be53ecaeb;rport
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=3576163364
To: <sip:+52xxxxxxxxxx@ims.telmex.com>
Call-ID: 80A6192D-2C81-E811-87B9-BB0BE53ECAEB@10.0.0.50
CSeq: 327 SUBSCRIBE
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>;+sip.instance="<urn:uuid:00D1773F-C27F-E811-97E2-EED72484AD05>"
Max-Forwards: 70
User-Agent: SIPPER for PhonerLite
Expires: 1800
Event: message-summary
Accept: application/simple-message-summary
Content-Length: 0
-------------------------------------------
09:52:17,705: R: 189.247.242.147:5060 (UDP)
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.50:5060;received=*.*.*.*;branch=z9hG4bK80a6192d2c81e81187bcbb0be53ecaeb;rport=60580
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=3576163364
To: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=aprqngfrt-cpkluc30080e4
Call-ID: 80A6192D-2C81-E811-87B9-BB0BE53ECAEB@10.0.0.50
CSeq: 327 SUBSCRIBE
-------------------------------------------
09:52:17,847: R: 189.247.242.147:5060 (UDP)
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.50:5060;received=*.*.*.*;branch=z9hG4bK80a6192d2c81e81187bbbb0be53ecaeb;rport=60580
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=1934202406
To: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=dqdq0bbq
Call-ID: 80A6192D-2C81-E811-87B8-BB0BE53ECAEB@10.0.0.50
CSeq: 326 REGISTER
WWW-Authenticate: Digest realm="ims.telmex.com", nonce="PBk3fi0dSOVla8Cwre8YRw==",algorithm=MD5
Content-Length: 0
-------------------------------------------
09:52:17,848: T: 189.247.242.147:5060 (UDP)
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.50:5060;branch=z9hG4bK80a6192d2c81e81187bdbb0be53ecaeb;rport
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=1934202406
To: <sip:+52xxxxxxxxxx@ims.telmex.com>
Call-ID: 80A6192D-2C81-E811-87B8-BB0BE53ECAEB@10.0.0.50
CSeq: 328 REGISTER
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>;+sip.instance="<urn:uuid:00D1773F-C27F-E811-97E2-EED72484AD05>"
Authorization: Digest username="+52xxxxxxxxxx@ims.telmex.com", realm="ims.telmex.com", nonce="PBk3fi0dSOVla8Cwre8YRw==", uri="sip:ims.telmex.com", response="bd636de3caf80c6c64ba2910c30555f9", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 900
Content-Length: 0
-------------------------------------------
09:52:18,100: R: 189.247.242.147:5060 (UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.50:5060;received=*.*.*.*;branch=z9hG4bK80a6192d2c81e81187bdbb0be53ecaeb;rport=60580
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=1934202406
To: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=k2dqaboc
Call-ID: 80A6192D-2C81-E811-87B8-BB0BE53ECAEB@10.0.0.50
CSeq: 328 REGISTER
P-Associated-URI: <sip:+52xxxxxxxxxx@ims.telmex.com;user=phone>
P-Associated-URI: <sip:+52xxxxxxxxxx@ims.telmex.com>
Accept-Resource-Priority: wps.4
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>;expires=30;q=1;+sip.instance="<urn:uuid:00D1773F-C27F-E811-97E2-EED72484AD05>"
Content-Length: 0
-------------------------------------------
09:52:18,123: T: 189.247.242.147:5060 (UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.247.242.147:5060;branch=z9hG4bKrc0icv204g1ge1u3epu0.1
From: <sip:+52xxxxxxxxxx@ims.telmex.com:5060>;tag=2gdrr79g-CC-20
To: <sip:+52xxxxxxxxxx@ims.telmex.com:60580>;tag=003db22d2c81e81187bdbb0be53ecaeb
Call-ID: 8g323bee3s8749i8a4r9r9iaa2a34s93@19500.0.ATS.ats01.ims.telmex.com.20
CSeq: 1 NOTIFY
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0
-------------------------------------------
09:52:18,293: R: 189.247.242.147:5060 (UDP)
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 10.0.0.50:5060;received=*.*.*.*;branch=z9hG4bK.bbkAMUUuC;rport=60580
From: <sip:52xxxxxxxxxx@voipnvcompigl.telmex.net>;tag=mTMk6w0eA
To: <sip:52xxxxxxxxxx@voipnvcompigl.telmex.net>;tag=uhf99zaa
CSeq: 210 REGISTER
Call-ID: grNrFpLzKe
Warning: 399 P.5.127.ims.telmex.com "SS170001F133L3261S0E0[00001] Hllm query failed"
Content-Length: 0
Yes, I’ve read it. The wiki has lots of info on FreePBX and DPMA - if you want to configure one phone at a time.
Dunno about EPM - I’m using DPMA because it’s supposed the Digium recommended method of configuration. I may have to look into EPM instead.
If it is ok with @tm1000 I will open a ticket for the issue on the tracker. I just was not sure if I should since I do some things differently in my guide, such as Not using Root user for install, and instead I modify asterisk.con and set the user and group to asterisk, I am also using Ubuntu instead of Debian. So I am doing a handful of things different from the norm here.
@tm1000 I was wondering if you could answer a couple things for me
asterisks included contrib/scripts/install_prereq install
takes care of a LOT of the dependencies.
I compared the dependencies it installs, with the ones listed on the wiki: https://wiki.freepbx.org/display/FOP/Installing+FreePBX+14+on+Debian+8.8
libsqlite3-dev and uuid-dev get installed by contrib/scripts/install_prereq install
So I am wondering if sqlite3 or uuid are really needed or if they can be omitted, I have installed without them and everything seems to works, but I figure I better ask since they are listed on the wiki
mongodb is also listed on the wiki, I was thinking maybe a feature or addon I am not using makes use of it, so I included it as well, but I would really like to know what it is used for.
I figured out nodejs is used for the UCP module.
pear install Console_Getopt, was already installed, without that line assuming thanks to contrib/scripts/install_prereq
I skipped iksemel and dahdi, pretty sure I do not need either of those.
Jansson was already installed by install_prereq as well.
Also I noticed in the wiki:
chown asterisk. /var/run/asterisk
chown -R asterisk. /var/{lib,log,spool}/asterisk
couldn’t that be changed into one line?
chown -R asterisk. /var/{lib,log,spool,run}/asterisk
or is there a reason to not do it that way? (The only difference is the -R recursive option, the only reason I can think of for not running it recursively on the run folder is there being something you would not want to affect?)
I noticed the Configure ODBC section of the wiki as well…
What happens if that section is skipped?
I did not have to do that section at all, and everything seems to be working (with the exception of having to run make samples)
To follow the original IPs mentioned above and not actual: when I had this (FreePBX generated conf file):
localnet=192.168.1.0/24
localnet=192.168.2.0/24
it didn’t work - but when I changed it to:
localnet=192.168.0.0/16
it did - two way audio for phones in both subnets
Both the prereq script and the FreePBX list are equally useful, consider them mutually additive, neither will upset the other and no packages would ever be uninstalled, so “it really doesn’t matter” . the asterisk script to ensure asterisk has all its pieces, the FreePBX list to ensure FPX has all its pieces. They are “mutually inclusive”
If you really want to be anal, you will spend quite a few hours and possibly save a few seconds off an install that you should only ever run once.
If I was you I would investigate why you are needlessly running ‘make samples’ . . . seriously . . . 
to quote innumerable devs from any named discipline, “Use the Source, Luke”
I’m having the exact same problem with the trunk and extension with the same IP address. Have you found a workaround?
Any ideas? Any suggestion please?
Possibly. In /etc/asterisk/pjsip_custom_post.conf try adding this line:
endpoint_identifier_order=auth_username,username,ip
This appears to cause PJSIP to try to match the auth_username and then username before the IP address (by default it matches the IP address first). It still won’t let you put an extension on a different port, but it does let you have a trunk and extension coming from the same IP address. For multiple extensions, maybe just the fact that they have different auth_usernames (their extension number) is sufficient in PJSIP even if they are using the same port, but I have not had the opportunity to test that. If you do, please let me know if it works.
I got some free time to do this now. The section is actually quite small. Going to try playing process of elimination with that section. I will just disable bit by bit of what it does, until I have it narrowed down.
will partition the drive and create a backup, so I can quickly restore, instead of having to start each install from square one.
Hey!
I upgraded the VPS that we are on to give us 100 GB of storage, vs. the 60 we were at previously. After upgrading the VPS, I booted an Ubuntu LiveCD, opened GParted, expanded the partition to fill the space, rebooted back into the FreePBX environment, and went from there.
Looking at system admin, it is reporting that I am still at 60GB total of storage, as is “df” (system admin “storage” screenshot attached)
Filesystem 1K-blocks Used Available Use% Mounted on
/dev/mapper/SangomaVG-root 56767404 41513932 15253472 74% /
devtmpfs 3992336 0 3992336 0% /dev
tmpfs 4004716 0 4004716 0% /dev/shm
tmpfs 4004716 8848 3995868 1% /run
tmpfs 4004716 0 4004716 0% /sys/fs/cgroup
/dev/vda1 1983056 201304 1662968 11% /boot
tmpfs 800944 0 800944 0% /run/user/995
tmpfs 800944 0 800944 0% /run/user/0
I feel like I’m missing a step, I just don’t know what. If needed, I can go back and provide a GParted GUI output to see if that helps at all.
https://linux.die.net/man/8/resize2fs
First make sure that you have a good backup, just in case.
I am very puzzled. I tried to replicate your issue (no response to REGISTER from Asterisk but ok from PhonerLite). I don’t have a Telmex account, so just tried +525522223333 and a dummy password.
Indeed, there was no response on my test system in Google Cloud. But unfortunately, there was also no response in PhonerLite. Possibly, requests with an invalid username are simply discarded.
It seems that you are not alone; see https://asteriskmx.org/foros/forum/asterisk/instalación-y-primeros-pasos/38921-configuracion-de-troncal-freepbx-telmex . Using that proxy, I did get a 403 with a Warning header showing “invalid user”, but don’t know whether that proxy would be valid for your account.
You posted in an old thread about a specific Italian provider. Try starting a new thread with Telmex in the title – with luck a Telmex user who has solved this will respond.
If not, one approach would be to copy a successful initial register to a file and send it with sipsak. Assuming that you get a response, gradually change it to what Asterisk was sending, one header at a time, testing after each change. Once you learn which header is the culprit, you should be able to change the Asterisk config to fix the problem.
Dear all,
I am trying to sync the caller name from other database to VoIP server.
I already success to update the name into asterisk database (tables: users, devices, sip).
the problem is, the calling name doesn’t display like the callerID that I synced. I already tried to reload the VoIP server (fwconsole reload).
when I tried “database show ampuser XXXX” the “/AMPUSER/XXXX/cidname” was not changed to the new callerID.
Thanks
This changes how matching is done globally for all endpoints, but you can also change it per endpoint.
For PJSIP trunks there is now a GUI option:
Setting this alone might actually solve the OP’s problem.
If not, identify_by=auth_username can be set for each endpoint in pjsip.endpoint_custom_post.conf.
[extension](+)
identify_by=auth_username