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9 participants
37392 discussions

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.*

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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

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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

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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.

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