- sr-users - kamailio.org

Re: [SR-Users] Load balance on HA scenario
by Yufei Tao 20 Oct '20

20 Oct '20

Hi Duarte, Not sure if I understand it correctly but are you trying to set up an HA pair of dispatchers? If so I am doing it differently in that, I use the Virtual/floating IP (VIP) as socket. This way the master node (which has got the VIP) can send OPTIONs pings out fine but the backup node won't be able to send. Functionally it is fine, but the backup node keeps logging errors because it doesn't have the VIP: ERROR: <core> [core/udp_server.c:599]: udp_send(): sendto(sock, buf: 0x7f0e6b54e2e0, len: 372, 0, dst: (192.168.213.143:5060), tolen: 16) - err: Network is unreachable (101) ERROR: tm [../../core/forward.h:231]: msg_send_buffer(): udp_send failed When FO happens, the backup node becomes master and takes the VIP, and it functions correctly. But I haven't worked out how to avoid the ERROR loggings on the backup node. Maybe someone else has any better ideas/suggestions? Cheers, Yufei

2 1

No retry on INVITE
by Allan Kristensen 20 Oct '20

20 Oct '20

Hello, I'm trying to get Kamailio to retry INVITE. We have a Freeswitch which is sending INVITE to a sip user via Kamailio proxy using static ip/location. All is well, calls are flowing, but sometimes INVITE from kamailio to user is lost and no attempt to retry is occurring and so after 30 sec kamailio just returns 408 Request timeout. Call flow is like this: FS Kamailio User --- INVITE --> <-- 100 Trying --- INVITE ---> (Waiting 30 secs) <-- 408 Timeout --- ACK ---> So assume we have failed in handling retries somehow. Kamailio is running stateless, but other retries are working, so I assume this is not a problem? Can someone perhaps point us in the right direction on where to solve this? Kamailio is version 5.2.4 Thank you... /Allan

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Re: [SR-Users] SRTP
by Yufei Tao 20 Oct '20

20 Oct '20

Hi, Like Olle said RTPEngine is perfectly capable of handling one leg in RTP the other in SRTP. When looking up RTPEngine's documentation, may be worth checking these out: SRTP SDES-unencrypted_srtp SDES-unencrypted_srtcp SDES-unauthenticated_srtp e.g. try apply these flags to rtpengin_offer() on the incoming RTP leg, to get RTPEngine to add encryption to the outgoing leg etc. Cheers, Yufei

1 0

Re: [SR-Users] Kamailio 5.2.5 - Need help with rtpengine error
by Yufei Tao 20 Oct '20

20 Oct '20

Hi Andrew, Like Alex said, have a look at the commands RTPEngine received and the responses it sent in the history of the call, and see if RTPEngine has ever received any commands regarding this call, or, if the call has been deleted from RTPEngine. You can raise the RTPEngine logging level to 6 for example so you can see the command/responses of RTPEngine. Cheers, Yufei

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about registrar save()/lookup()
by Juha Heinanen 20 Oct '20

20 Oct '20

I didn't find any mention in registrar module README that save() function saves also branch flags and that lookup() retrieves them. Is that documented somewhere? -- Juha

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dialog_active_dialogs erroneous value
by Sergiu Pojoga 20 Oct '20

20 Oct '20

Hi there, Every now and then, the value of active dialogs metric spikes to an erroneous huge number, messing up my otherwise beautiful grafana gauges :( Example: kamailio_dialog_active_dialogs 18446744073709551614 1602812239115 kamailio_dialog_early_dialogs 0 1602812239115 kamailio_dialog_expired_dialogs 2 1602812239115 kamailio_dialog_failed_dialogs 7361 1602812239115 Anybody faced the same symptoms, any clues what might be causing this? Thanks, --Sergiu

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TOPOS module and fix_nated_contact
by Minh Phan 19 Oct '20

19 Oct '20

Hi List, I am using Topos module to hide network topology. I am having a problem of routing the ACK request for REINVITE. Here is the schema: A KAMAILIO B 1. --------- INVITE --------> 2. <---- 180 RINGING --- 3. <---- 200 OK --- 4. ---- ACK ---> 5. <--------- (RE) INVITE -- 6. --------- OK ---> 7. <---- ACK ----- (THIS ACK IS SENT TO PRIVATE IP OF A) I use TOPOS module along with fix_nated_contact as I would do normally for natted UAs. The first ACK (4) is routed correctly but not the second one (7), which is routed to the PRIVATE IP of A instead of routing to its public address. Has anyone run into this problem while using TOPOS module? How could I make them work together? Best Regards, Minh Phan

2 1

hold_on_music over P2P call [Kamailio]
by Amit Pal 19 Oct '20

19 Oct '20

Hi Team, Pls guide me on following point. :- By using kamailio (P2P call), want to play hold on music during hold call. Shall it be possible without any external media server like (RTPProxy/media proxy). Regards Amit Pal

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Kamailio IMS RTP Engine
by Pavithra Mohanraja 19 Oct '20

19 Oct '20

Hi, I have followed below link to setup rtp engine. https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ RTP engine is started without any errors and it is enabled in pcscf also. I am using the config files of pcscf as per the above link .. still rtp packets are not reaching pcscf from UE . What could be the problem? The only difference is that i have deployed ims in kubernetes environment as pods. Kindly help. Thanks, Pavithra

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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

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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

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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

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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

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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

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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

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Error in RTP engine setup
by Pavithra Mohanraja 15 Oct '20

15 Oct '20

Hi, I tried to setup rtpengine to make my sip clients communicate through proxy following https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ I started sip communication since the service was running and saw that the RTP packets go to exposed rtp interface instead of other sip client. But i don't receive packets from the other end. When analyzed I noticed the rtp-engine daemon gives a few errors. *modprobe: FATAL: Module xt_RTPENGINE not found in directory /lib/modules/4.15.0-118-generic* *iptables: No chain/target/match by that name.ip6tables: No chain/target/match by that name. * Starting RTP/media proxy ngcp-rtpengine-daemon [1602690253.130392] ERR: FAILED TO CREATE KERNEL TABLE 0 (No such file or directory), KERNEL FORWARDING DISABLED* *(Note: *uname -r gives 4.15.0-118-generic and /lib/modules/ has only 4.15.0-115-generic*)* Please help me with this issue. Thanks in Advance. Regards, Pavithra

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Re: [SR-Users] Regd : SIP Configuration in Kamailio
by Henning Westerholt 14 Oct '20

14 Oct '20

Hello, please keep the list in CC. I can see Kamailio in your log: Oct 14 11:21:59 nickar /usr/sbin/kamailio[24567]: INFO: <core> [main.c:842]: sig_usr(): signal 15 received Oct 14 11:21:59 nickar systemd[1]: Stopping Kamailio (OpenSER) - the Open Source SIP Server... Oct 14 11:21:59 nickar /usr/sbin/kamailio[24568]: INFO: <core> [main.c:842]: sig_usr(): signal 15 received About tracing/debugging, one good option is sngrep. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Santanu B <santanu.br(a)gmail.com> Sent: Wednesday, October 14, 2020 1:41 PM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio Hi, My syslog file is enclosed for your reference. Thanks, Santanu On Wed, Oct 14, 2020 at 1:07 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello, have a look to /var/log/messages or /var/log/syslog, depending on the distribution. Kamailio might be logged there. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Santanu B Sent: Wednesday, October 14, 2020 7:09 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio Dear David, In /var/log folder I am unable to see any kamailio log. Only syslog file is available. Please help me to find the Kamailio log. Also please guide me to find out the trace also as I am very new to SIP Server. Thanks a lot. Regards, Santanu On Tue, Oct 13, 2020 at 4:05 PM David Villasmil <david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com>> wrote: Hello, You will need to provide more information. What happens? What errors do you see in kamailio’s log? A trace would also be nice. On Tue, 13 Oct 2020 at 06:24, Santanu B <santanu.br(a)gmail.com<mailto:santanu.br@gmail.com>> wrote: 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 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com> phone: +34669448337 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

SRTP
by Andy Kama 14 Oct '20

14 Oct '20

Hi All, I have setup a tls connection to a sip provider and it works perfectly But they are requesting i send srtp i can find any documentation how i do it with rtpengine etc? any guidance would be appreciated

3 3

pstn_route add_route(): route name <IP> not defined error
by sai Sudheer 14 Oct '20

14 Oct '20

Hii,I am trying to use pstn_route module for the first time, and i want to route the invite to specific pstn server based on its prefix.I have loaded the module & setup the DB, and it's giving error saying route name (from DB) is not defined after adding the a prefix into DB table., and there were no errors when DB table is empty.Not sure how to defined it, or i dont know what did i miss. Please help. #### CONFIG ##########loadmodule "prefix_route.so"modparam("prefix_route", "db_url", "mysql://xxxxx:xxxxx@localhost/kamailioDB")modparam("prefix_route", "db_table", "new_prefix_route") #PSTN Routing table contains if (!prefix_route("+44")) { xlog("L_INFO", "+44 prefix didnt match with prefix_route DB\n"); } ########## DB####MariaDB [kamailio]> select * from new_prefix_route;Empty set (0.00 sec) MariaDB [kamailio]> exit;########## Kamailio looks good untill here, no errors.. # Now i have added a prefix into sql DBINSERT INTO new_prefix_route VALUES ("+44", "10.10.1.1:5060", "route for this prefix");###########Error log after adding a prefix into new_prefix_route table & restart of kamailio 0(5802) CRITICAL: prefix_route [prefix_route.c:72]: add_route(): route name '10.10.1.1:5060' is not defined 0(5802) NOTICE: prefix_route [prefix_route.c:170]: pr_db_load(): Total prefix routes loaded: 1 0(5802) ERROR: prefix_route [prefix_route.c:174]: pr_db_load(): error flushing tree 0(5802) CRITICAL: prefix_route [prefix_route.c:212]: mod_init(): db load failed I have tried a "name" instead of IP (though i dont know how to convert that "name" into a IP:Port), But i am still getting same error using "name" instead of IP:port @ route field in DB. Please help.

2 2

how to get a value from SIP INVITE Header
by sai sudheer 14 Oct '20

14 Oct '20

Hii Please help me to get the a value from SIP INVITE header reached to kamailio like INVITE sip:+341930203454@sub.domain.com;myid=+34@sub.domain.com SIP/2.0 and i want to save the myid value +34 into a variable, without the domain name. $var(uri) = $sel(ruri); xavp_params_explode("$(var(uri){s.unbracket})", "uri"); xlog("L_INFO", "$var(uri) Received converted to $xavp(uri=>myid[0])\n"); I tried above and it prints +34(a)sub.domain.com But i want to just save +34 into a variable to further check the prefix based routing from the database. 1) Could you please help how to get it or If there is any alternate/single line approach to get this value? 2) $var & $xavp are process-local variables, and they cant be shared with other calls? Right 3) What is the best module to route calls based their prefix ? pstn_route or dynamic routing or any suggestions? Please help, thanks in advance.

3 7

'302 Moved temporarily' contact parsing error
by David VILLAUME 14 Oct '20

14 Oct '20

Hello, I see parsing errors on 302 contact header that seems compliant to RFC : Contact:<sip:172.16.0.21:5060>;q=0.5,<sip:172.16.0.22:5060>;q=0.25 <core> [core/parser/parse_addr_spec.c:479]: parse_to_param(): invalid character ',' in status 27: [;q=0.5] Is there an error in my contact or is it truly a parsing error ? Regards, David

3 6

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