sr-users May 2015

sr-users@lists.kamailio.org

80 participants
115 discussions

Forward register to multiple Asterisk
by Nelson Migliaro 01 Jun '15

01 Jun '15

Hello, I am trying to forward the register to multiple Asterisk using Kamailio. The basic Idea is: Softphone A---->Kamailio-----> Asterisk 1 ------> Asterisk 2. I follow Asipto howto http://lylix.net/kamailio, but the problem I see in my case, is that registration IP in Asterisk is Kamailio´s IP. This causes calls going from kamailio to asterisk and then back to kamailio. In my scenario, I need that softphone A IP appeard in Asterisk realtime contact IP. The call path would be: Softphone B -> kamailio -> Asterisk 1 or Asterisk 2 (depending on dispatcher) -> Softphone A Following Asipto how to the call goes Softphone A -> kamailio -> asterisk -> kamailio Softphone B Hope I explain myself clearly. Thank you in advance.

2 1

Kamailio , RTPProxy and NAT
by Mir Jee 01 Jun '15

01 Jun '15

HI! I need help to configure RTPProxy. I have working Kamailio on Centos 6.5, On LAN everything is working fine. Now I want my other users to connect using 3G or any other network. Please help me with that. Statis Public IP (DSL Modem) ----- Proxy Box with 2 Lan cards --- 192.168.2.0 and 192.168.0.0 (Lan Side)

2 1

Kamailio 300 Multiple choices
by Ali Taher 29 May '15

29 May '15

Hello, We need to integrate Kamailio with an exchange switch in order to have routing decision. Kamailio must send to the switch the list of available gateways in the contact tag of SIP response as shown in the below example: SIP/2.0 300 Multiple choices Via: SIP/2.0/UDP 172.16.5.112:5060;branch=z9hG4bK1sansay9906915rdb2243 Via: SIP/2.0/UDP 216.53.4.5:5060;branch=z9hG4bK_1852370774_4741_1 To: <sip:86532917453592@69.41.186.186:5060 <http://sip:86532917453592@69.41.186.186:5060> > From: <sip:+249964642906@216.53.4.5:5060;otg=024050>;tag=1852370774_C Call-ID: sbcsipuac.2_169.132.137.51_b55sb12_1_1_2015052909455960_1852370774_487698 CSeq: 1 INVITE Contact: <sip:2917453592@036310>, <sip:2917453592@016500>, <sip:138295#2917453592@044610>, <sip:2917453592@020100> Server: Invoice SIP-server Content-Length: 0 Is this can be done through drouting module , or we need to use another module ? Your help is highly appreciated. Regards, Ali

2 7

"Multiple Phone Numbers" registration
by Al S 29 May '15

29 May '15

Hi, Does Kamilio have a predefined module to support "Multiple Phone Numbers" registration with a single Registration?I am trying to register multiple numbers on behalf of IP-PBXs. Thanks,Al

2 3

hash table not loading
by Olli Attila 29 May '15

29 May '15

Hello, I am trying to load a test hash table on kamailio startup but I keep getting the following error with kamctl after i restart Kamailio and try to list all the tables in memory: kamcmd> htable.listTables error: 500 - No htables I have the following configured in kamailio.cfg: # ---- loading the htable module loadmodule "htable.so" # ----- htable params ----- modparam("htable", "htable", "abc=>size=8;autoexpire=0;dbtable=htable;") modparam("htable", "db_url", DBURL) modparam("htable", "fetch_rows", 1000) The DBURL is pointed to the kamailio database on the same machine. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif I have also added the following values for testing purposes to the kamailio.htable: id key_name key_type value_type key_value expires ------ ---------- -------- ---------- --------- --------- 1 test 0 0 12345 0 Is there anything else that i might be missing here? Im running kamailio v.4.2.4. Cheers, -- Olli

3 8

Kamailio V4.2.5 append prefix to Request Uri problem.
by Yasin CANER 29 May '15

29 May '15

Hello; i try to append prefix to $rU like below in Kamailio V4.2.5 . i get error even if i set string value. it works only when i append like that $rU="3002"+$rU; --------- 1 test------------ $var(prefix)=$avp(s:69); $rU=$var(prefix)+$rU; -FAILED ---------2 test ------------ $rU=$avp(s:69)+$rU; -FAILED ------- 3 test ------------ $rU=$avp(i:69)+$rU; -FAILED -------ERROR--------- May 29 12:06:57 group1-imslab11 /usr/local/sbin/kamailio[26196]: ERROR: pv [pv_core.c:2115]: pv_set_ruri_user(): str value required to set R-URI user May 29 12:06:57 group1-imslab11 /usr/local/sbin/kamailio[26196]: ERROR: <core> [lvalue.c:363]: lval_pvar_assign(): setting pvar failed May 29 12:06:57 group1-imslab11 /usr/local/sbin/kamailio[26196]: ERROR: <core> [lvalue.c:416]: lval_assign(): assignment failed at pos: (790,7-790,22) it work on Version 4.1.7 like below correctly; $rU=$avp(i:69)+$rU; Thanks Yasin

3 3

Kamailio users creation mechanism
by Alexandru Covalschi 29 May '15

29 May '15

Hello! My goal is to create a mechanism for dynamic creation/deletion of SIP users on Kamailio. First question: is it possible to create users not via kamctl command, but directly inserting new entries to this tables? And the second one: if it is possible, which hash alghorythm Kamailio uses when creating a new user on subscriber table? Becase I see 2 fields with hashes. Thank you -- Alexandru Covalschi VoIP engineer and system administrator phone: +37367398493 <http://abs-telecom.com/>

2 1

Elastix MT (3.0) issue with NAT (ACK is not received for 200 OK)
by Mainul Haque 28 May '15

28 May '15

Hello. I think I understand the problem, but I do not know how to fix it. I have done the following when the problem accord: -Download Elastix MT 64bit (downloaded from - http://www.elastix.com/en/downloads/) -Install it in VMware ESXi 5.1. Elastix works fine within the LAN -Then I pushed the VM to EC2 by exporting VMware OVF -I noticed that I could not registered my softphone (eyebeam 1.5) -After doing research online, I had to modify /etc/kamailio/kamailio-mhomed-elastix.cfg to know about the new IP of EC2 instance. See below of what I had to change: Old settings: [root@new-host-2 endpoint_configurator]# grep 192 /etc/kamailio/kamailio-mhomed-elastix.cfg if (is_in_subnet($var(target_remote_ip), "192.168.1.0/24")) { $var(rtpproxy_if) = "192.168.1.62"; $var(rtpproxy_if) = "192.168.1.6"; To New Settings: [root@new-host-2 ~]# grep 172 /etc/kamailio/kamailio-mhomed-elastix.cfg if (is_in_subnet($var(target_remote_ip), "172.31.48.0/18")) { $var(rtpproxy_if) = "172.31.55.101"; $var(rtpproxy_if) = "172.31.55.101"; -Now I can register my eyebeam softphone. -When ever I dial anything, the connection would *drop after 10 to 30 seconds.* -When I enable SIP debug, I get the following: ############################################################# [May 28 16:07:58] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> INVITE sip:*65@voiptosave.com SIP/2.0 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Record-Route: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.3cc1dc7f932835a200ec1f8feafd0895.0 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-182127345f1d7b4f-1--d87543-;rport=29470 Max-Forwards: 69 Contact: <sip:110@192.168.1.6:29470> To: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080> From: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 457 v=0 o=- 2 2 IN IP4 192.168.1.62 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.62 t=0 0 m=audio 18598 RTP/AVP 100 106 6 0 105 8 18 3 5 101 a=alt:1 1 : KjAnzMl+ lOdYVIyR 192.168.1.6 15880 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:60C3A67CD0E643D8A6945053F94B1D3E a=nortpproxy:yes <-------------> [May 28 16:07:58] VERBOSE[2688] chan_sip.c: --- (15 headers 17 lines) --- [May 28 16:07:58] VERBOSE[2688] chan_sip.c: Sending to 127.0.0.1:5060 (NAT) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Sending to 127.0.0.1:5060 (NAT) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Using INVITE request as basis request - Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found peer ' 110_voiptosave.com' for '110_voiptosave.com' from 127.0.0.1:5060 [May 28 16:07:58] VERBOSE[2688][C-00000021] netsock2.c: == Using SIP RTP TOS bits 184 [May 28 16:07:58] VERBOSE[2688][C-00000021] netsock2.c: == Using SIP RTP CoS mark 5 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 100 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 106 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 6 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 0 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 105 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 8 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 18 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 3 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 5 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 101 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found audio description format SPEEX for ID 100 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found unknown media description format SPEEX-FEC for ID 106 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found unknown media description format SPEEX-FEC for ID 105 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found audio description format G729 for ID 18 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found audio description format telephone-event for ID 101 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw|alaw|adpcm|g729|speex16)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Peer audio RTP is at port 192.168.1.62:18598 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Looking for *65 in voiptosave.com-from-internal (domain voiptosave.com) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: list_route: hop: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: list_route: hop: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.3cc1dc7f932835a200ec1f8feafd0895.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-182127345f1d7b4f-1--d87543-;rport=29470 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Record-Route: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> From: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e To: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080> Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 INVITE Server: Asterisk PBX 11.13.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:*65@127.0.0.1:5080> Content-Length: 0 <------------> [May 28 16:07:58] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:1] Answer("SIP/110_voiptosave.com-0000001a", "") in new stack [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Audio is at 17370 [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Adding codec 100003 (ulaw) to SDP [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Adding codec 100004 (alaw) to SDP [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Adding codec 100002 (gsm) to SDP [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: <--- Reliably Transmitting (NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.3cc1dc7f932835a200ec1f8feafd0895.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-182127345f1d7b4f-1--d87543-;rport=29470 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Record-Route: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> From: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e To: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 INVITE Server: Asterisk PBX 11.13.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:*65@127.0.0.1:5080> Content-Type: application/sdp Content-Length: 275 v=0 o=root 153043023 153043023 IN IP4 127.0.0.1 s=Asterisk PBX 11.13.0 c=IN IP4 127.0.0.1 t=0 0 m=audio 17370 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [May 28 16:07:58] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:2] Wait("SIP/110_voiptosave.com-0000001a", "1") in new stack [May 28 16:07:58] VERBOSE[2688] chan_sip.c: Retransmitting #1 (NAT) to 127.0.0.1:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.3cc1dc7f932835a200ec1f8feafd0895.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-182127345f1d7b4f-1--d87543-;rport=29470 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Record-Route: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> From: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e To: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 INVITE Server: Asterisk PBX 11.13.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:*65@127.0.0.1:5080> Content-Type: application/sdp Content-Length: 275 v=0 o=root 153043023 153043023 IN IP4 127.0.0.1 s=Asterisk PBX 11.13.0 c=IN IP4 127.0.0.1 t=0 0 m=audio 17370 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [May 28 16:07:58] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> ACK sip:*65@127.0.0.1:5080 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.1131e459ea9f93c78896b0fdaf99b3be.0 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-9f74bc13426dd510-1--d87543-;rport=29470 Max-Forwards: 69 Contact: <sip:110@192.168.1.6:29470> To: "*65"<sip:*65@voiptosave.com>;tag=as3a404cf6 From: "Manny"<sip:110@voiptosave.com>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 ACK Proxy-Authorization: Digest username="110",realm="voiptosave.com ",nonce="VWd2ylVndZ5Yo3kaAdjLh/W6zc+CeoE1",uri="sip:*65@voiptosave.com ",response="ed5ad3dd093d74414340aa68dca6a246",algorithm=MD5 User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 0 <-------------> [May 28 16:07:58] VERBOSE[2688] chan_sip.c: --- (12 headers 0 lines) --- [May 28 16:07:58] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> ACK sip:*65@127.0.0.1:5080 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.1131e459ea9f93c78896b0fdaf99b3be.0 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-9f74bc13426dd510-1--d87543-;rport=29470 Max-Forwards: 69 Contact: <sip:110@192.168.1.6:29470> To: "*65"<sip:*65@voiptosave.com>;tag=as3a404cf6 From: "Manny"<sip:110@voiptosave.com>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 ACK Proxy-Authorization: Digest username="110",realm="voiptosave.com ",nonce="VWd2ylVndZ5Yo3kaAdjLh/W6zc+CeoE1",uri="sip:*65@voiptosave.com ",response="ed5ad3dd093d74414340aa68dca6a246",algorithm=MD5 User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 0 <-------------> [May 28 16:07:58] VERBOSE[2688] chan_sip.c: --- (12 headers 0 lines) --- [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:3] Macro("SIP/110_voiptosave.com-0000001a", "voiptosave.com-user-callerid,") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:1] Set("SIP/110_voiptosave.com-0000001a", "EXTUSER=110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:2] GotoIf("SIP/110_voiptosave.com-0000001a", "0?report") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:3] ExecIf("SIP/110_voiptosave.com-0000001a", "1?Set(REALCALLERIDNUM=110)") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:4] Set("SIP/110_voiptosave.com-0000001a", "EXTUSER=110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:5] Set("SIP/110_voiptosave.com-0000001a", "EXTUSERCIDNAME=User_110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:6] GotoIf("SIP/110_voiptosave.com-0000001a", "0?report") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:7] Set("SIP/110_voiptosave.com-0000001a", "EXTUSERCID=110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:8] Set("SIP/110_voiptosave.com-0000001a", "CALLERID(all)="User_110" <110>") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:9] ExecIf("SIP/110_voiptosave.com-0000001a", "1?Set(CHANNEL(language)=en)") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:10] ExecIf("SIP/110_voiptosave.com-0000001a", "1?Set(CHANNEL(language)=en)") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:11] GotoIf("SIP/110_voiptosave.com-0000001a", "0?continue") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:12] Set("SIP/110_voiptosave.com-0000001a", "__TTL=64") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:13] GotoIf("SIP/110_voiptosave.com-0000001a", "1?continue") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-user-callerid,s,20) [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:20] Set("SIP/110_voiptosave.com-0000001a", "CALLERID(number)=110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:21] Set("SIP/110_voiptosave.com-0000001a", "CALLERID(name)=User_110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:22] NoOp("SIP/110_voiptosave.com-0000001a", "Using CallerID "User_110" <110>") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:4] Playback("SIP/110_voiptosave.com-0000001a", "your") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'your.gsm' (language 'en') [May 28 16:08:00] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:5] Playback("SIP/110_voiptosave.com-0000001a", "extension") in new stack [May 28 16:08:00] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'extension.gsm' (language 'en') [May 28 16:08:01] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:6] Playback("SIP/110_voiptosave.com-0000001a", "number") in new stack [May 28 16:08:01] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'number.gsm' (language 'en') [May 28 16:08:02] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:7] Playback("SIP/110_voiptosave.com-0000001a", "is") in new stack [May 28 16:08:02] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'is.gsm' (language 'en') [May 28 16:08:03] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:8] SayDigits("SIP/110_voiptosave.com-0000001a", "110") in new stack [May 28 16:08:03] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'digits/1.gsm' (language 'en') [May 28 16:08:03] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'digits/1.gsm' (language 'en') [May 28 16:08:04] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'digits/0.gsm' (language 'en') [May 28 16:08:05] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:9] Wait("SIP/110_voiptosave.com-0000001a", "2") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:10] Hangup("SIP/110_voiptosave.com-0000001a", "") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: == Spawn extension (voiptosave.com-from-internal, *65, 10) exited non-zero on 'SIP/110_voiptosave.com-0000001a' [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [h@voiptosave.com-from-internal:1] Macro("SIP/110_voiptosave.com-0000001a", "voiptosave.com-hangupcall") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:1] ExecIf("SIP/110_voiptosave.com-0000001a", "1?Set(CDR(organization_domain)= voiptosave.com)") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:2] GotoIf("SIP/110_voiptosave.com-0000001a", "1?endmixmoncheck") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,7) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:7] NoOp("SIP/110_voiptosave.com-0000001a", "End of MIXMON check") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:8] GotoIf("SIP/110_voiptosave.com-0000001a", "1?nomeetmemon") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,14) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:14] NoOp("SIP/110_voiptosave.com-0000001a", "MEETME_RECORDINGFILE=") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:15] GotoIf("SIP/110_voiptosave.com-0000001a", "1?noautomon") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,23) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:23] NoOp("SIP/110_voiptosave.com-0000001a", "TOUCH_MONITOR_OUTPUT=") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:24] GotoIf("SIP/110_voiptosave.com-0000001a", "1?noautomon2") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,31) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:31] NoOp("SIP/110_voiptosave.com-0000001a", "MONITOR_FILENAME=") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:32] GotoIf("SIP/110_voiptosave.com-0000001a", "1?skiprg") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,35) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:35] GotoIf("SIP/110_voiptosave.com-0000001a", "1?skipblkvm") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,38) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:38] GotoIf("SIP/110_voiptosave.com-0000001a", "1?theend") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,40) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:40] Hangup("SIP/110_voiptosave.com-0000001a", "") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] app_macro.c: == Spawn extension (macro-voiptosave.com-hangupcall, s, 40) exited non-zero on 'SIP/110_voiptosave.com-0000001a' in macro 'voiptosave.com-hangupcall' [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: == Spawn extension (voiptosave.com-from-internal, h, 1) exited non-zero on 'SIP/110_voiptosave.com-0000001a' [May 28 16:08:07] VERBOSE[9110][C-00000021] chan_sip.c: Scheduling destruction of SIP dialog 'Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA.' in 6400 ms (Method: ACK) [May 28 16:08:07] VERBOSE[9110][C-00000021] chan_sip.c: set_destination: Parsing <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> for address/port to send to [May 28 16:08:07] VERBOSE[9110][C-00000021] chan_sip.c: set_destination: set destination to 127.0.0.1:5060 [May 28 16:08:07] VERBOSE[9110][C-00000021] chan_sip.c: Reliably Transmitting (NAT) to 127.0.0.1:5060: BYE sip:110@192.168.1.6:29470 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2355deaa;rport Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes>,<sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Max-Forwards: 70 From: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 To: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 102 BYE User-Agent: Asterisk PBX 11.13.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [May 28 16:08:07] VERBOSE[2688] chan_sip.c: Retransmitting #1 (NAT) to 127.0.0.1:5060: BYE sip:110@192.168.1.6:29470 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2355deaa;rport Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes>,<sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Max-Forwards: 70 From: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 To: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 102 BYE User-Agent: Asterisk PBX 11.13.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [May 28 16:08:07] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2355deaa;rport=5080 Contact: <sip:110@192.168.1.6:29470> To: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e From: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 102 BYE User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 0 <-------------> [May 28 16:08:07] VERBOSE[2688] chan_sip.c: --- (9 headers 0 lines) --- [May 28 16:08:07] VERBOSE[2688][C-00000021] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [May 28 16:08:07] VERBOSE[2688] chan_sip.c: Really destroying SIP dialog 'Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA.' Method: ACK [May 28 16:08:08] VERBOSE[2688] chan_sip.c: Reliably Transmitting (NAT) to 127.0.0.1:5060: OPTIONS sip:110@192.168.1.6:29470;rinstance=3180b40a89f2e3f4 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK0bcd981c;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as68358641 To: <sip:110@192.168.1.6:29470;rinstance=3180b40a89f2e3f4> Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 03d50e985a2f08445ad315c83a591412@127.0.0.1:5080 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.13.0 Date: Thu, 28 May 2015 20:08:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [May 28 16:08:08] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK0bcd981c;rport=5080 Contact: <sip:192.168.1.6:29470> To: <sip:110@192.168.1.6:29470;rinstance=3180b40a89f2e3f4>;tag=86756729 From: "asterisk"<sip:asterisk@127.0.0.1:5080>;tag=as68358641 Call-ID: 03d50e985a2f08445ad315c83a591412@127.0.0.1:5080 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 0 <-------------> [May 28 16:08:08] VERBOSE[2688] chan_sip.c: --- (12 headers 0 lines) --- [May 28 16:08:08] VERBOSE[2688] chan_sip.c: Really destroying SIP dialog ' 03d50e985a2f08445ad315c83a591412@127.0.0.1:5080' Method: OPTIONS ################################################################## Please let me know if you need any settings. I am more than happy to provide it. Thank you.

1 1

log the value of "from_uri" and "myself"
by Al S 28 May '15

28 May '15

Hi, I would like to know how to log the value of "from_uri" and "myself" attributes on kamailio 3.3.7 ?None of the following works: xlog("L_INFO","from_uri: $dlg(from_uri)"); $var(x) = from_uri;xlog("L_INFO","$var(x)"); Thanks,Al

2 3

Kamailio not receiving BYE request
by Ali Taher 28 May '15

28 May '15

Hello, Kamailio is not receiving the BYE request when the call is ended by any of call parties. Noting that when the call is connected, a direct RTP session is established between the user and the gateway , and kamailio doesn't receive since then any message related to this call. Is there any trick to do in order to let kamailio gets the BYE request? Attached is the used configuration. Appreciate your help. Regards, Ali

3 11

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sr-users May 2015