Unsupported trunk not working

[marzo82]

we have 1 trunk ISP Telmex, but when we try to call or out call to this trunk not work, this is the error:
09/30/2021 12:36:15 PM - Malformed SIP request received:
BadSip Recv Req INVITE from 10.148.0.16:52868 tid=C9691877 [email protected]:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.148.0.16:5060;branch=z9hG4bKC9691877
Max-Forwards: 65
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: "528127243680 " <sip:[email protected]>;tag=585D9034-D92
Call-ID: [email protected]
CSeq: 101 INVITE
Expires: 180
Session-Expires: 1800
Min-SE: 1800
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Content-Disposition: session;handling=required
Content-Type: application/sdp
Date:
Supported: 100rel, timer, resource-priority, replaces, sdp-anat
Timestamp: 1633023363
User-Agent: Cisco-SIPGateway/IOS-15.7.3.M
Allow-Events: telephone-event
Cisco-Guid: 3082591613-0558567916-2220737817-1554932602
Session-ID: 6bc5f819ad05528ca9bcf244a9e66da8;remote=00000000000000000000000000000000
Content-Length: 267

v=0
o=CiscoSystemsSIP-GW-UserAgent 5052 2671 IN IP4 10.148.0.16
s=SIP Call
c=IN IP4 10.148.0.16
t=0 0
m=audio 16574 RTP/AVP 18 101
c=IN IP4 10.148.0.16
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

any idea to resolve this.

thanks
regards

 

[YiannisH_3CX]

Hi @marzo82

We will need more details about your setup and the way the provider is setup to work in your system to be able to assist. Start with the information as described here: https://www.3cx.com/community/threads/information-to-provide-when-requesting-help.67558/

Also as the provider you are using is not a supported on try a supported provider and see if that works for you

 

[Mayank143]

Dear Mr.Marzo82,


Kindly try this & remember that for each total dialing digits (10/12/16/20...etc) you have to make separate outbound rules.

DISABLED SRTP & set Transport Protocol as ANY + ADD / DEL the CODECs which approved by SIP provider, then try below outbound rule, if successful then try to change SRTP & Transport protocol but be sure that your SIP really authorized/work on this.

For Outbound calls, you should make the Outbound rules simply then it would work, as you made it already so here I'm re-writing which maybe re-gain you if you missed something:-
1. Calls to numbers starting with prefix - 00

2. Calls from extension(s) - MENTION THE EXTENSION OR if more then add with comma BUT NO SPACE

3. Calls to Numbers with a length of - EXACT LENGTH, IF MOBILE NUMBER IS "0091 1234567890" = 14DIGITS SO MUST MENTION 14, IF DIALING LANDLINE NUMBER OR ANY OTHER COUNTRY WHICH MAYBE 12 OR 16 OR MORE THEN MAKE A SEPARATE OUTBOUND RULE FOR EACH AS one rule for one type of dialing. Each Outbound rule will auto-detect, don't worry.

4. Calls from extension group(s) - leave it OR if you made DEFAULT group & Want to give the access to all then add the same group.

5. Make outbound calls on - In the Route 1 choose SIP name then choose 0 in Strip Digits, leave Prepend blank, choose Outbound Caller ID

Dial like this 0091 1234567890 where 00 triggers the OUTBOUND RULE & as an International call + total 14digit trigger the exact length for this rule.

Save it & try...

B.regd
Mayank Gupta