Issue : Not getting relay of ACK and BYE to the next hop after the call is answered my Scenario : Asterisk ------->kamailio sip proxy-------------------> carrier (outgoing call)
My carrier is not allowed to send the SIP packet with Record-Route header. So that I have removed record_route(). After that the call is getting connected. I am getting 200 OK (SDP) from carrier side and forwarded that to the Asterisk on the other side. As a response I am getting ACK from asterisk. But the kamailio is not forwarding the ACK to the carrier side. I understood this is because the record-route is not there. The same thing is happening for BYE also. The Bye is not forwarding to carrier side.
Kindly suggest me a solution for this for relaying ACK and bye without Record-Route in kamailio
Bellow is the 200 OK SDP I am sending back to asterisk 2024/06/02 10:27:04.756610 103.155.114.101:5060 -> 103.182.153.113:5060 SIP/2.0 200 OK Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 Via: SIP/2.0/UDP 103.182.153.113:5060 ;received=103.182.153.113;rport=5060;branch=z9hG4bKPj463fedcf-6258-4655-a53d-58ea57f144af To: <sip:09496381412@103.155.114.101
;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109
From: <sip:917946357720@gaesip.teleforce.in
;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1
CSeq: 22823 INVITE Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE Contact: sip:lucentNGFS-115804@10.5.110.117:5060 ;alias=10.5.110.117~5060~1;x-afi=11 Content-Type: application/sdp Session-Expires: 7200;refresher=uas Supported: timer Content-Length: 248
v=0 o=LucentPCSF 130227946 130227946 IN IP4 103.155.114.101 s=- c=IN IP4 103.155.114.101 t=0 0 m=audio 12806 RTP/AVP 8 101 -------------------------------------------------------------------------------------------------------
The ACK I am getting back from asterisk is 2024/06/02 10:27:04.760392 103.182.153.113:5060 -> 103.155.114.101:5060 ACK sip:lucentNGFS-115804@103.155.114.101:5060;alias=10.5.110.117~5060~1;x-afi=11 SIP/2.0 Via: SIP/2.0/UDP 103.182.153.113:5060 ;rport;branch=z9hG4bKPjb7299c25-13d3-484e-8e78-e7c83620edce From: <sip:917946357720@gaesip.teleforce.in
;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1
To: <sip:09496381412@103.155.114.101
;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109
Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 CSeq: 22823 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 18.13.0 Content-Length: 0
Thanks Arun
As mentioned above - use TOPOS BUT - return record_route() back.
On Sun, Jun 2, 2024, 08:40 Nick Digalakis via sr-users < sr-users@lists.kamailio.org> wrote:
Hi!
Check out the TOPOS module.
It does topology hiding and it should do what you want.
On Jun 2, 2024 08:13, Arun K R via sr-users sr-users@lists.kamailio.org wrote:
Issue : Not getting relay of ACK and BYE to the next hop after the call is answered my Scenario : Asterisk ------->kamailio sip proxy-------------------> carrier (outgoing call)
My carrier is not allowed to send the SIP packet with Record-Route header. So that I have removed record_route(). After that the call is getting connected. I am getting 200 OK (SDP) from carrier side and forwarded that to the Asterisk on the other side. As a response I am getting ACK from asterisk. But the kamailio is not forwarding the ACK to the carrier side. I understood this is because the record-route is not there. The same thing is happening for BYE also. The Bye is not forwarding to carrier side.
Kindly suggest me a solution for this for relaying ACK and bye without Record-Route in kamailio
Bellow is the 200 OK SDP I am sending back to asterisk 2024/06/02 10:27:04.756610 103.155.114.101:5060 -> 103.182.153.113:5060 SIP/2.0 200 OK Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 Via: SIP/2.0/UDP 103.182.153.113:5060 ;received=103.182.153.113;rport=5060;branch=z9hG4bKPj463fedcf-6258-4655-a53d-58ea57f144af To: <sip:09496381412@103.155.114.101
;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109
From: <sip:917946357720@gaesip.teleforce.in
;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1
CSeq: 22823 INVITE Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE Contact: sip:lucentNGFS-115804@10.5.110.117:5060 ;alias=10.5.110.117~5060~1;x-afi=11 Content-Type: application/sdp Session-Expires: 7200;refresher=uas Supported: timer Content-Length: 248
v=0 o=LucentPCSF 130227946 130227946 IN IP4 103.155.114.101 s=- c=IN IP4 103.155.114.101 t=0 0 m=audio 12806 RTP/AVP 8 101
The ACK I am getting back from asterisk is 2024/06/02 10:27:04.760392 103.182.153.113:5060 -> 103.155.114.101:5060 ACK sip:lucentNGFS-115804@103.155.114.101:5060;alias=10.5.110.117~5060~1;x-afi=11 SIP/2.0 Via: SIP/2.0/UDP 103.182.153.113:5060 ;rport;branch=z9hG4bKPjb7299c25-13d3-484e-8e78-e7c83620edce From: <sip:917946357720@gaesip.teleforce.in
;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1
To: <sip:09496381412@103.155.114.101
;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109
Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 CSeq: 22823 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 18.13.0 Content-Length: 0
Thanks Arun
Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Hello, Thank you for your response, Could you please tell me what do you mean by return record_route() back ? I have implemented topos module and now call is working with record_route(). But now for the BYE I am getting 403 GLU unmatched from the carrier. Could you suggest something for this
Thanks again Arun
On Sun, Jun 2, 2024 at 5:36 PM Yuriy G ovoshlook@gmail.com wrote:
As mentioned above - use TOPOS BUT - return record_route() back.
On Sun, Jun 2, 2024, 08:40 Nick Digalakis via sr-users < sr-users@lists.kamailio.org> wrote:
Hi!
Check out the TOPOS module.
It does topology hiding and it should do what you want.
On Jun 2, 2024 08:13, Arun K R via sr-users sr-users@lists.kamailio.org wrote:
Issue : Not getting relay of ACK and BYE to the next hop after the call is answered my Scenario : Asterisk ------->kamailio sip proxy-------------------> carrier (outgoing call)
My carrier is not allowed to send the SIP packet with Record-Route header. So that I have removed record_route(). After that the call is getting connected. I am getting 200 OK (SDP) from carrier side and forwarded that to the Asterisk on the other side. As a response I am getting ACK from asterisk. But the kamailio is not forwarding the ACK to the carrier side. I understood this is because the record-route is not there. The same thing is happening for BYE also. The Bye is not forwarding to carrier side.
Kindly suggest me a solution for this for relaying ACK and bye without Record-Route in kamailio
Bellow is the 200 OK SDP I am sending back to asterisk 2024/06/02 10:27:04.756610 103.155.114.101:5060 -> 103.182.153.113:5060 SIP/2.0 200 OK Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 Via: SIP/2.0/UDP 103.182.153.113:5060 ;received=103.182.153.113;rport=5060;branch=z9hG4bKPj463fedcf-6258-4655-a53d-58ea57f144af To: <sip:09496381412@103.155.114.101
;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109
From: <sip:917946357720@gaesip.teleforce.in
;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1
CSeq: 22823 INVITE Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE Contact: sip:lucentNGFS-115804@10.5.110.117:5060 ;alias=10.5.110.117~5060~1;x-afi=11 Content-Type: application/sdp Session-Expires: 7200;refresher=uas Supported: timer Content-Length: 248
v=0 o=LucentPCSF 130227946 130227946 IN IP4 103.155.114.101 s=- c=IN IP4 103.155.114.101 t=0 0 m=audio 12806 RTP/AVP 8 101
The ACK I am getting back from asterisk is 2024/06/02 10:27:04.760392 103.182.153.113:5060 -> 103.155.114.101:5060 ACK sip:lucentNGFS-115804@103.155.114.101:5060;alias=10.5.110.117~5060~1;x-afi=11 SIP/2.0 Via: SIP/2.0/UDP 103.182.153.113:5060 ;rport;branch=z9hG4bKPjb7299c25-13d3-484e-8e78-e7c83620edce From: <sip:917946357720@gaesip.teleforce.in
;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1
To: <sip:09496381412@103.155.114.101
;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109
Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 CSeq: 22823 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 18.13.0 Content-Length: 0
Thanks Arun
Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
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Arun K R -
Nick Digalakis -
Yuriy G