sr-users October 2020

sr-users@lists.kamailio.org

66 participants
84 discussions

Libphonenumber - Updating metadata
by Zach Nielsen 18 Oct '20

18 Oct '20

Hey all, I've been using the PHONENUM module, which uses Google's libphonenumber library, to perform number validation among other handy things within the route script. One problem I've run into however is the metadata in the latest stable (5.4) repo is pretty stale. I'm curious if anyone is running this module, and if so how they've gone about updating the metadata used by the underlying library. If I have to ditch the repo and compile everything manually, that's not a huge deal. Just hoping there's another way. Thanks! Zach

3 3

No trusted table found
by Jon Bonilla (Manwe) 17 Oct '20

17 Oct '20

Hi all Just installed a kamailio 5.4.1 for a simple proxy scenario. I created the version,address and trusted tables, and added some ips to the trusted via kamctl. If I execute "kamctl trusted show" I see the list of servers. But with "kamctl trusted dump" I receive { "jsonrpc": "2.0", "error": { "code": 500, "message": "No trusted table" }, "id": 13063 } when I start kamailio with debug I see these messages: b_do_submit_query(): submitted query: select `table_version` from `version` where `table_name`='trusted', result 0 my version table looks like this: +----+------------+---------------+ | id | table_name | table_version | +----+------------+---------------+ | 1 | version | 1 | | 2 | trusted | 6 | | 3 | address | 6 | +----+------------+---------------+ any hints? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 5

TLS connection from Chrome to Kamailio fails
by Juha Heinanen 17 Oct '20

17 Oct '20

White testing WebRTC, I noticed that TLS connection from Chrome browser 85.0.4183.121 to Kamailio's (master version) TLS listening port fails. Chrome sends Client Hello and K responds with Server Hello. After that Chrome terminates the handshake by sending TCP FIN. Connection setup from Firefox works fine. After receiving Server Hello, is sends Change Cipher Spec, etc. My K is using Let's Encrypt server certificate and is server tls_method is TLSv1.2+. Any ideas what goes wrong? Has anyone else experienced this? -- Juha

3 6

How to increase private & shared memory in kamailio Container?
by Hamid Hashmi 16 Oct '20

16 Oct '20

Dear Users How can I increase private memory in the Kamailio container? the service was started with the following command docker run -d --restart unless-stopped --network host -v /root/volumes/scscf/:/etc/kamailio --name scscf -e IP=10.161.224.90 -e INT_IP=10.161.224.90 10.161.224.196:5000/kamailio-ngvoice PS: I don't have a docker file to build the image again. Regards Hamid R. Hashmi

2 1

usrloc_dmq - Suddenly stopped usrloc replication
by José Seabra 16 Oct '20

16 Oct '20

Hello there, Suddenly the usrloc replication through DMQ stopped, but the interesting thing is that DMQ cluster status is OK, htable replication still working only usrloc_dmq stopped working. I increased the kamailio loglevel to 3 in order to see if there was some log related and then registered a peer, but the only DMQ logs that I see are related with notification_peer and htable replication, nothing related with usrloc_dmq. What I can do more here to try to understand what went wrong? I haven't restarted the cluster yet, because I would like to try to understand the root cause of this issue. Any advice would be appreciated Thank you -- Cumprimentos José Seabra

3 8

Kamailio 5.2.5 - Need help with rtpengine error
by Andrew Chen 15 Oct '20

15 Oct '20

Hi all, Today I saw the following errors reported by rtpengine module: Oct 15 15:02:10 ashprodkama52 kamailio[8028]: 30(8086) DEBUG: rtpengine [rtpengine.c:2453]: rtpp_function_call(): proxy reply: d6:result5:error12:error-reason15:Unknown call-ide Oct 15 15:02:10 ashprodkama52 kamailio[8028]: 30(8086) ERROR: rtpengine [rtpengine.c:2474]: rtpp_function_call(): proxy replied with error: Unknown call-id This was on the 200 OK response back from our Freeswitch. Any idea what would cause this? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Regd : SIP Configuration in Kamailio
by Santanu B 15 Oct '20

15 Oct '20

Hi, We are using Jitsi Meet for Video Conference Service. We want our SIP endpoint to join in the conference room in Jitsi. For that i am trying to configure Kamailio SIP Server in our environment. I have installed the Server and my endpoints are registered in my SIP server. Both endpoints are able to connect to each other. But I am unable to configure the SIP endpoints to join in the Meet Conference Room. Please guide me to configure. Thanks and regards, Santanu

5 5

sdpops - modifying sdp codecs' presedence list - order
by Melek Oktay 15 Oct '20

15 Oct '20

Hello, Kamailio *sdpops *module consist of several useful functions for manipulating SDP on Proxy side such as *sdp_keep_codecs_by_name*() . This function *sdp_keep_codecs_by_name*("PCMU,PCMA,GSM") finds codecs id [0,8,3] and if incoming packet consist of other than these codecs, they are removed ! For example, assume incoming packet SDP --> [8 0 3 9 101] after applying this function 9 and 101 are removed and outgoing packet SDP consist only PCMU,PCMA,GSM [0,8,3]. This function does taking given codec *precedence/priority* into account. As you see above message SDP is 8 0 3, different priority than inputted 0 8 3. I need to change *order *of codecs too ! Would you help me to do it ? I couldn't find *add *operation on *sdpops *module Regards, Melek

1 0

Kamailio Relay Replies Issue
by me.projects＠yahoo.com 15 Oct '20

15 Oct '20

I have DID provider which is sending INVITES to Kamailio and kamailio has UAC module with asterisk to forward Calls to Asterisk with uac_auth() 1-DID Provider ----> (TLS)2-Kamailio (UDP) -----> (UDP) 3-Asterisk Call from Kamailio to Asterisk is forwarded and answered but NON of the reply received in "MANAGE_REPLY" route is not relayed back to DID provider call get disconnected. =================SIP DUMP INVITE FROM DID INVITE sip:+1700@KAMAILIOSERER.com:5061;user=phone;transport=tls SIP/2.0FROM: Alga Tech<sip:+17900000002@DIDPROVIDER.com:5061;user=phone>;tag=28989e96e28d47d891b4d8d21d85fa92TO: <sip:+1700@KAMAILIOSERER.com:5061;user=phone>CSEQ: 1 INVITECALL-ID: 936ba28f302f5a5d9b09f32e2e7c41f2MAX-FORWARDS: 70VIA: SIP/2.0/TLS DID-IP:5061;TH=div;branch=z9hG4bK77976715RECORD-ROUTE: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>CONTACT: <sip:DIDPROVIDER-CONTACT.com:443;x-i=b9483d0b-e3aa-4487-8dec-68de62d93a35;x-c=936ba28f302f5a5d9b09f32e2e7c41f2/d/8/26d69f6b729942a799552629a315d5d1>CONTENT-LENGTH: 1103MIN-SE: 300SUPPORTED: timerUSER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.10.12.5 i.USEA.0CONTENT-TYPE: application/sdpALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFYSESSION-EXPIRES: 3600TH: dih -----------------------Invite to Asterisk with AUTH INVITE sip:+1700@KAMAILIOSERER.com:5061;user=phone;transport=tls SIP/2.0FROM: Alga Tech<sip:+17900000002@DIDPROVIDER.com:5061;user=phone>;tag=28989e96e28d47d891b4d8d21d85fa92TO: <sip:+1700@KAMAILIOSERER.com:5061;user=phone>CSEQ: 1 INVITECALL-ID: 936ba28f302f5a5d9b09f32e2e7c41f2MAX-FORWARDS: 69VIA: SIP/2.0/TLS DID-IP:5061;TH=div;branch=z9hG4bK77976715RECORD-ROUTE: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>CONTACT: <sip:DIDPROVIDER-CONTACT.com:443;x-i=b9483d0b-e3aa-4487-8dec-68de62d93a35;x-c=936ba28f302f5a5d9b09f32e2e7c41f2/d/8/26d69f6b729942a799552629a315d5d1>CONTENT-LENGTH: 1103MIN-SE: 300SUPPORTED: timerUSER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.10.12.5 i.USEA.0CONTENT-TYPE: application/sdpALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFYSESSION-EXPIRES: 3600TH: dih ------------------ >From Asterisk to Kamailio SIP/2.0 100 TryingVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>CSeq: 1 INVITEServer: AsteriskContent-Length: 0TH: uch ---------------------- >From Asterisk to Kamailio SIP/2.0 180 RingingVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq: 1 INVITEServer: AsteriskContact: <sip:66.232.107.106:7790>Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContent-Length: 0 ---------------------- >From Asterisk to Kamailio SIP/2.0 180 RingingVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq: 1 INVITEServer: AsteriskAllow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContact: <sip:66.232.107.106:7790>Content-Length: 0TH: uch ---------------------- >From Asterisk to Kamailio SIP/2.0 200 OKVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq: 1 INVITEServer: AsteriskAllow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContact: <sip:66.232.107.106:7790>Supported: 100rel, timer, replaces, norefersubSession-Expires: 3600;refresher=uacRequire: timerContent-Type: application/sdpContent-Length: 299TH: uch

1 0

To Header Manipulation
by Sam Ware 15 Oct '20

15 Oct '20

Wondering if this is intended behavior. We have a customer doing some craziness with the TO header. They are including their tech prefix in the TO header as well as the RURI. It doesn't cause any issue in our systems but apparently it does with one of our vendors. In an effect to prevent the issue, we thought we would check if the tech prefix was added on the TO header and remove it. Initially, I tried this by changing the $tU variable. $avp(tp_len) = $(avp(techprefix){s.len}) + 1; # Get the Length of the Tech Prefix and add 1 for the # or * $tU = $(tU{s.strip, $avp(tp_len)}); The result ended up with the duplication of the DNIS: sut <sip:911999# 1812555111118125551111@172.16.3.45:5060> Where 911999# was the tech prefix and 18125551111 is the DNIS which was still an issue for the vendor. Through additional reading, I found the suggestion to use the uac_replace_to function in the UAC module which I implemented as follows: $avp(new_to_hdr_uri) = $(tu{re.subst,/sip:(.*[\*#])(.*)/sip:\2/}); uac_replace_to("$avp(new_to_hdr_uri)"); This resulted in the original TO header username to be appended at the end of the uri: sut <sip:18125551111@172.16.3.45:5060911999#18125551111> Wondering if I am doing something wrong or this is the way these functions are designed to protect the TO header contents for Dialog matching. -- Sam D Ware

2 1

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sr-users October 2020