Forward DID to SIP trunk
Re: Forward DID to SIP trunk
Hi,
From the tcpdump traces, what I understand the dialed number (My cell no for example) is considered as an SIP extension in the OXE, actually this call should go out to the Telco provider.
Im attaching a snippet (tcp dump).
xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
13:58:56.190570 IP (tos 0x0, ttl 63, id 3369, offset 0, flags [none], proto: UDP (17), length: 903) 10.40.13.252.5060 > main.5060: SIP, length: 875
INVITE sip:0813035xxxx@10.41.12.203 SIP/2.0
Via: SIP/2.0/UDP 10.40.13.252:5060;branch=z9hG4bK6569d863
Max-Forwards: 70
From: "kmrl" <sip:88888@10.40.13.252>;tag=as02a84234
To: <sip:0813035xxxx@10.41.12.203>
Contact: <sip:88888@10.40.13.252:5060>
Call-ID: 33c566c36eb6578a3db5a35972b8c601@10.40.13.252:5060
CSeq: 102 INVITE
User-Agent: IPBX-2.11.0(16.16.1)
Date: Sat, 05 Mar 2022 08:28:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 299
v=0
o=root 1563111237 1563111237 IN IP4 10.40.13.252
s=Asterisk PBX 16.16.1
c=IN IP4 10.40.13.252
t=0 0
m=audio 11714 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
13:58:56.196083 IP (tos 0x0, ttl 63, id 3373, offset 0, flags [none], proto: UDP (17), length: 445) 10.40.13.252.5060 > main.5060: SIP, length: 417
ACK sip:0813035xxxx@10.41.12.203 SIP/2.0
Via: SIP/2.0/UDP 10.40.13.252:5060;branch=z9hG4bK6569d863
Max-Forwards: 70
From: "kmrl" <sip:88888@10.40.13.252>;tag=as02a84234
To: <sip:0813035xxxx@10.41.12.203>;tag=89113b6275e53f27bae3db686038d511
Contact: <sip:88888@10.40.13.252:5060>
Call-ID: 33c566c36eb6578a3db5a35972b8c601@10.40.13.252:5060
CSeq: 102 ACK
User-Agent: IPBX-2.11.0(16.16.1)
Content-Length: 0
xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
10.40.13.252 - Asterisk
10.41.12.203 - OXE
813035xxxx - Cell no (10 digit)
I need to dial "0" from my OXE, the PRI connection gets through then cell number with prefix "0" , for normal calling (Call from OXE extension).
I need this from Asterisk.
regards
anoop
From the tcpdump traces, what I understand the dialed number (My cell no for example) is considered as an SIP extension in the OXE, actually this call should go out to the Telco provider.
Im attaching a snippet (tcp dump).
xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
13:58:56.190570 IP (tos 0x0, ttl 63, id 3369, offset 0, flags [none], proto: UDP (17), length: 903) 10.40.13.252.5060 > main.5060: SIP, length: 875
INVITE sip:0813035xxxx@10.41.12.203 SIP/2.0
Via: SIP/2.0/UDP 10.40.13.252:5060;branch=z9hG4bK6569d863
Max-Forwards: 70
From: "kmrl" <sip:88888@10.40.13.252>;tag=as02a84234
To: <sip:0813035xxxx@10.41.12.203>
Contact: <sip:88888@10.40.13.252:5060>
Call-ID: 33c566c36eb6578a3db5a35972b8c601@10.40.13.252:5060
CSeq: 102 INVITE
User-Agent: IPBX-2.11.0(16.16.1)
Date: Sat, 05 Mar 2022 08:28:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 299
v=0
o=root 1563111237 1563111237 IN IP4 10.40.13.252
s=Asterisk PBX 16.16.1
c=IN IP4 10.40.13.252
t=0 0
m=audio 11714 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
13:58:56.196083 IP (tos 0x0, ttl 63, id 3373, offset 0, flags [none], proto: UDP (17), length: 445) 10.40.13.252.5060 > main.5060: SIP, length: 417
ACK sip:0813035xxxx@10.41.12.203 SIP/2.0
Via: SIP/2.0/UDP 10.40.13.252:5060;branch=z9hG4bK6569d863
Max-Forwards: 70
From: "kmrl" <sip:88888@10.40.13.252>;tag=as02a84234
To: <sip:0813035xxxx@10.41.12.203>;tag=89113b6275e53f27bae3db686038d511
Contact: <sip:88888@10.40.13.252:5060>
Call-ID: 33c566c36eb6578a3db5a35972b8c601@10.40.13.252:5060
CSeq: 102 ACK
User-Agent: IPBX-2.11.0(16.16.1)
Content-Length: 0
xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
10.40.13.252 - Asterisk
10.41.12.203 - OXE
813035xxxx - Cell no (10 digit)
I need to dial "0" from my OXE, the PRI connection gets through then cell number with prefix "0" , for normal calling (Call from OXE extension).
I need this from Asterisk.
regards
anoop
Re: Forward DID to SIP trunk
If you whant to send 0813035xxxx from SIP to PRI.
The most simple way:
- send 00813035xxxx from Asterisk (duble 0 in the begining - I think 0 - seize prefix in OXE)
- in SIP TG manage Transcoding= false (or use NPD with DID translator in transparent mode without rules)
- check COS/Connection COS 5 (1 in position 5).
- check Entity/ discriminator selector (in SIP TG Entity shall be used the same discriminator for ISDN TG like in user Entity) or use in SIP TG the same Entity like for OXE users
OXE will receive 00813035xxxx
Without transcoding number will be used as is 0 prefix for ISDN TG (I hope) and numer 0813035xxxx will be send via ISDN TG.
The most simple way:
- send 00813035xxxx from Asterisk (duble 0 in the begining - I think 0 - seize prefix in OXE)
- in SIP TG manage Transcoding= false (or use NPD with DID translator in transparent mode without rules)
- check COS/Connection COS 5 (1 in position 5).
- check Entity/ discriminator selector (in SIP TG Entity shall be used the same discriminator for ISDN TG like in user Entity) or use in SIP TG the same Entity like for OXE users
OXE will receive 00813035xxxx
Without transcoding number will be used as is 0 prefix for ISDN TG (I hope) and numer 0813035xxxx will be send via ISDN TG.
Re: Forward DID to SIP trunk
Hello Mr Vad,
Let me try, will let you know the result
regards
anoop
Let me try, will let you know the result
regards
anoop
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- Member
- Posts: 49
- Joined: 25 Nov 2013 08:26
Re: Forward DID to SIP trunk
Hello
I also need help.....about DID Cli issue on incomming call....on another oxe node...
f.e. isdn DID pri terminated on oxe node 1.... All did of node 1 working perfect incoming and outgoing cli ok....but....DID of node 2 incoming call cli not display on set...outgoing cli is ok....
Node 1 and node 2 connected with ABCF configuration...
I also need help.....about DID Cli issue on incomming call....on another oxe node...
f.e. isdn DID pri terminated on oxe node 1.... All did of node 1 working perfect incoming and outgoing cli ok....but....DID of node 2 incoming call cli not display on set...outgoing cli is ok....
Node 1 and node 2 connected with ABCF configuration...
Re: Forward DID to SIP trunk
ok put some t3 traces for that case
-
- Member
- Posts: 49
- Joined: 25 Nov 2013 08:26
Re: Forward DID to SIP trunk
hi haroun
our site faraway ,and internet issue....discuss with site optr...calling no and called no are displayed in t3 traces...in both nodes...Let me know if you have any tips...thanks
our site faraway ,and internet issue....discuss with site optr...calling no and called no are displayed in t3 traces...in both nodes...Let me know if you have any tips...thanks
Re: Forward DID to SIP trunk
external callback translator managed on node 2?
how users on node §2 are called from node 1?
how users on node §2 are called from node 1?
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- Member
- Posts: 49
- Joined: 25 Nov 2013 08:26
Re: Forward DID to SIP trunk
user call from node 1 to node 2 through prefix plan= routing no...and prefix plan= network no.
second (external callback translator managed on node 2?) Does this not apply to outgoing calls?(reverse dialing)
second (external callback translator managed on node 2?) Does this not apply to outgoing calls?(reverse dialing)
Re: Forward DID to SIP trunk
yep external call back translator is needed for clisadi_seemi wrote: ↑09 Nov 2022 02:37 user call from node 1 to node 2 through prefix plan= routing no...and prefix plan= network no.
second (external callback translator managed on node 2?) Does this not apply to outgoing calls?(reverse dialing)
Re: Forward DID to SIP trunk
can u make a test call and put the t3 taces