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Setup Trunk without authentication (Vodafone IP)

July 15, 2019, 3:19 am
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I activated pjsip debug and I’m getting this messages:
There is one error " SIP/2.0 400 Bad Request" “Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)”

Any Idea whats wrong here?

<— History Entry 3580 Received from 88.79.204.9:5060 at 1563185367 —>
INVITE sip:+49LOCALNUMBER@pbx1.nucleus.ngn.vodafone.de:5060 SIP/2.0
Via: SIP/2.0/UDP 88.79.204.9:5060;received=88.79.204.9;branch=z9hG4bKebq34h00983et2ekci50.1;origin=172.19.116.80
To: sip:+49LOCALNUMBER@nbgsx001.ngn.vodafone.de;user=phone
From: sip:+49EXTERNALNUMBER@ims.vodafone.de;user=phone;tag=SD0r0b001-2239645f
Call-ID: SD0r0b001-c395cdbe96ad36f5ddb5dfbfa3ea4f60-ct4u830040
CSeq: 1 INVITE
Max-Forwards: 60
Contact: sip:+49EXTERNALNUMBER@88.79.204.9:5060;transport=udp
Date: Mon, 15 Jul 2019 12:09:27 GMT
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Supported: resource-priority
P-Asserted-Identity: sip:+49EXTERNALNUMBER@ims.vodafone.de;user=phone
P-Asserted-Identity: tel:+49EXTERNALNUMBER
Accept: application/sdp
P-Early-Media: supported
Content-Type: application/sdp
Content-Length: 289
Content-Type: application/sdp
Content-Length: 289

v=0
o=- 0 0 IN IP4 88.79.204.9
s=IMSS
c=IN IP4 88.79.204.9
t=0 0
m=audio 55000 RTP/AVP 96 9 8 101 102
a=rtpmap:101 telephone-event/8000
a=rtpmap:102 telephone-event/16000
a=ptime:20
a=maxptime:30
a=fmtp:96 mode-set=0,1,2; mode-change-period=2; mode-change-neighbor=1; max-red=0

<— History Entry 3581 Sent to 88.79.204.9:5060 at 1563185367 —>
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 88.79.204.9:5060;rport=5060;received=88.79.204.9;branch=z9hG4bKebq34h00983et2ekci50.1;origin=172.19.116.80
Call-ID: SD0r0b001-c395cdbe96ad36f5ddb5dfbfa3ea4f60-ct4u830040
From: sip:+49EXTERNALNUMBER@ims.vodafone.de;user=phone;tag=SD0r0b001-2239645f
To: sip:+49LOCALNUMBER@nbgsx001.ngn.vodafone.de;user=phone;tag=z9hG4bKebq34h00983et2ekci50.1
CSeq: 1 INVITE
Warning: 399 SIP “Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)”
Server: FPBX-14.0.13.4(15.4.0)
Content-Length: 0

<— History Entry 3582 Received from 88.79.204.9:5060 at 1563185367 —>
ACK sip:+49LOCALNUMBER@pbx1.nucleus.ngn.vodafone.de:5060 SIP/2.0
Via: SIP/2.0/UDP 88.79.204.9:5060;received=88.79.204.9;branch=z9hG4bKebq34h00983et2ekci50.1;origin=172.19.116.80
CSeq: 1 ACK
To: sip:+49LOCALNUMBER@nbgsx001.ngn.vodafone.de;user=phone;tag=z9hG4bKebq34h00983et2ekci50.1
From: sip:+49EXTERNALNUMBER@ims.vodafone.de;user=phone;tag=SD0r0b001-2239645f
Call-ID: SD0r0b001-c395cdbe96ad36f5ddb5dfbfa3ea4f60-ct4u830040
Max-Forwards: 60
Content-Length: 0
Content-Length: 0

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Using FaxPRO, fax received, but not attachment in the email

July 15, 2019, 3:57 am
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Force IVR to ignore invalid user entries

July 15, 2019, 4:52 am
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You cannot have your cake and eat it too with this. You have two options 1) Set the invalid destination timeouts really high. You won’t get the announcement but the IVR will just continue to wait for valid entries until that very high limit is reached. 2) You set the invalid retry to 1 and the invalid destination to the IVR itself. This will mean that every invalid entry gets the invalid entry playback and will restart the IVR at the greeting again to let them choose other options.

You’re not going to get the invalid announcement just to play and not do anything else. Just not going to happen.

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UCP EPM (Device Management) Apply button does nothing

July 15, 2019, 5:19 am
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Outbound CID ignored

July 15, 2019, 5:50 am
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Hey guys,

when I call an external number via sip-trunk I want to transmit my external-no and not my internal extension-no.
Therefor I set “outbound cid” in my user-settings to my external number.
Unfortunately the internal extension-no is used.

To be honest I’m a bit shocked that this is even possible… I call an external no. and they are receiving a call from “13” ??? How is this even possible…?

Sip-Log:

<— History Entry 14974 Sent to 88.79.204.9:5060 at 1563195076 —>
INVITE sip:EXTERNALNO@88.79.204.9:5060 SIP/2.0
Via: SIP/2.0/UDP 92.79.124.70:5060;rport;branch=z9hG4bKPje3085dd5-4af6-484b-9050-62b0a31680f8
From: “MYNAME” sip:13@MYDOMAIN.ngn.vodafone.de;tag=da7e7a0e-a180-44db-a46a-2b29ebecc284
To: sip:EXTERNALNO@88.79.204.9
Contact: sip:asterisk@MYIP:5060
Call-ID: f790530c-fe42-411e-88f8-f2ac5d7a87d1
CSeq: 8187 INVITE
Route: sip:88.79.204.9:5060;lr
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: “MYNAME” sip:13@MYDOMAIN.ngn.vodafone.de
Remote-Party-ID: “MYNAME” sip:13@MYDOMAIN.ngn.vodafone.de;privacy=off;screen=no
Privacy: none
P-Preferred-Identity: sip:MYEXTERNALNO@MYIP;user=phone
Max-Forwards: 70
User-Agent: FPBX-14.0.13.4(15.4.0)
Content-Type: application/sdp
Content-Length: 306

v=0
o=- 697628934 697628934 IN IP4 92.79.124.70
s=Asterisk
c=IN IP4 MYIP
t=0 0
m=audio 45612 RTP/AVP 9 8 0 3 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

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Errors in crontab file, can't install - No changes can be saved

July 15, 2019, 6:06 am
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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

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Cisco 7942 , SIP/2.0 401 Unauthorized

July 15, 2019, 6:08 am
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Thank you all of your Reply
I tried alot but failed.
Finally request to my vendor change IP Phone model .

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Cisco 7942 , SIP/2.0 401 Unauthorized

July 15, 2019, 6:12 am
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Considering we actually never saw a full debug we couldn’t determine what the actual problem is. Like I said, there’s always going to be an initial 401 Unauthorized challenge to the first INVITE attempt. That’s all you’ve shown in your original post. That is normal behavior which means we need to see more of it. Like the INVITE with the auth details being passed in it and the reply to that message.

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Mitel Phones Unavailable/On The Phone

July 15, 2019, 6:14 am
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In my experience, this sounds like your keep-alive time is set too high for your system to keep the phones connected. The fact that they drop off and ‘drop back on’ seemingly at random tells me that the problem has got to be something in the way the configurations for your phones and PBX are just not meshing.

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Getting error TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 27 PJSIP

July 15, 2019, 6:15 am
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What are you trying to change, and how are you trying to change it.

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Cisco 7942 , SIP/2.0 401 Unauthorized

July 15, 2019, 6:16 am
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@BlazeStudios just seen only logs which i posted on my question.
Not seen any of debug log beside that.

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FreePBX complains of tampered file?

July 15, 2019, 6:23 am
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… or a failing memory module somewhere in the system. There are several levels of memory in the average system, and you could be seeing the failure of a cache module, a drive controller cache, on on-disk memory buffer, or a bad hard drive itself.

The ‘fire and forget’ way to solve it is to replace the system wholesale and image the drive onto a new drive. Imaging the drive onto a new drive would probably be a good idea anyway, since problems like this (once they start) usually only get worse.

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Outbound CID ignored

July 15, 2019, 6:28 am
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Outbound CID ignored

July 15, 2019, 6:38 am
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I see this all the time when I set the outbound trunk to “Intracompany Route”. I’ll bet you have that turned on.

You can also override the extension’s number in the outbound by setting the outbound trunk to the Caller ID you want to use and set it to “ignore extensions settings”.

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Who's working on FreePBX? (for pay)

July 15, 2019, 7:36 am
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On the forum over the past few months we have seen announcements about FreePBX staff leaving the company, but I have not seen much if anything from remaining or new staff who work on FreePBX.

I realize this is a community forum and maybe we have been spoiled by all the great posts here from @xrobau, @tm1000, @jfinstrom, and @tonyclewis over the years (sorry if I missed someone). And of course @lgaetz is still with us, both in terms of being on the staff and contributing in the forum. So who’s left / new? I think I speak for more than just myself in saying we’d like to hear more from you.

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(Unofficial) FreePBX on Pi

July 15, 2019, 7:53 am
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If that’s Ron’s script, that compiles asterisk. That’s a terrible idea because then you don’t have all the QA resources of the distro you’re using managing security updates.

Now, some people may WANT to compile their own asterisk, and they may have some odd reason to do so, but 99% of people don’t (even if they think they do!)

The script I posted takes about 2 or 3 minutes to run, depending on your network speed.

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FreePBX complains of tampered file?

July 15, 2019, 7:54 am
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I find it unlikely it will be RAM, because RAM would only corrupt the file when it’s being written - it wouldn’t corrupt a file that’s at rest.

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(Unofficial) FreePBX on Pi

July 15, 2019, 7:55 am
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100% this.

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Permission issue when SSH as non-root

July 15, 2019, 8:02 am
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Just had this happen today on my company PBX and I did not run any module updates or anything since that last time I was in the system on Friday.

So some normal process must cause this to happen.

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Control multiple Inbound Routes with Single Flow Control

July 15, 2019, 8:02 am
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I’m trying to find a solution for the following situation;

  • There are multiple DID’s, each configured with his own inbound route.
  • Each inbound route has a different destination (IVR, Ring Group , etc)

I want to change the destination of every DID with a single Flow Control.
There are solution where you trigger multiple Flow Controls from a single Flow Control, but that is not the solution I need… (you will need n+1 Flow Controls, and is doubles with every new flow…)

This is the normal way to set-up the flow control;

Inbound Routes:

  • DID: 666-6666666 (Destination: Flow Control FC6)
  • DID: 777-7777777 (Destination: Flow Control FC7)
  • DID: 888-8888888 (Destination: Flow Control FC8)

Flow Control:
FC6 with;

  • Day(6) to Ring Group (6)
  • Night(6) to Announcement (6)
    FC7 with;
  • Day(6) to Ring Group (7)
  • Night(6) to Announcement (7)
    FC8 with;
  • Day(6) to Ring Group (8)
  • Night(6) to Announcement (8)

In this situation you have to change 3 Flow Controls (or you uses some code to change those 3 from a single FC

I’m trying to create a different way of handling Centralized Flow Control;

Inbound Routes:

  • DID: 666-6666666 (Destination: Central Flow Control)

  • DID: 666-66666661 (Destination: Ring Group 6)

  • DID: 666-66666662 (Destination: Announcement 6)

  • DID: 777-7777777 (Destination: Central Flow Control)

  • DID: 777-77777771 (Destination: Ring Group 7)

  • DID: 777-77777772 (Destination: Announcement 7)

  • DID: 888-8888888 (Destination: Central Flow Control)

  • DID: 888-88888881 (Destination: Ring Group 8)

  • DID: 888-88888882 (Destination: Announcement 8)

Flow Control:

  • Day: Misc Destination : *9991 (custom feature code 999 with option 1)
  • Night : Misc Destination : *9992 (custom feature code 999 with option 2)

When someone calls: 777-7777777…
Flow Control (Day) will: use Misc Destination *9991

“Some” code will: place call in {from-trunk} and add “1” to the DID

Inbound Route DID: 777-77777771 will be used and go to Destination: Ring Group 7

I've modified extensions_custom.conf, and added;
======================================================
[ext-featurecodes-custom]
exten => *9991,1,Goto(from-internal,*9991,1)
;--== end of [ext-featurecodes-custom] ==--;

[Modified-DID]
exten => *9991,1,Goto(from-trunk,${EXTEN}1,1)
;--== end of [Modified-DID] ==--;
======================================================

This does not work…

Does someone know how to direct a call to the inbound routes, and add a digit to the DID?

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