- sr-users - kamailio.org

SIP Auth Realm - From domain or Request domain ?
by jay binks 04 Jun '15

04 Jun '15

I have a questions in regards to SIP Auth, and specifically how Asterisk 13 with PJSIP appears to send Invites. It seems that the typical Kamailio config sends the SIP auth challenge with the from domain as the auth realm. however in the case of the Asterisk 13 ( PJSIP ) invite, the from domain is an RFC1918 IP address and I would have to use the domain from the request. What im wanting to know is what the consensus is on where the auth realm comes from. does everyone 100% agree that you send auth using the SIP From header, or is this something that some do Request domain ? What im considering doing is to use the from domain, but if from domain = RFC1918 then use request domain. Im just after everyone's opinion on how to best handle this invite, from a auth_challenge perspective. ( not so much if the Asterisk 13 PJSIP module is doing the right or wrong thing ) INVITE sip:555111222333@sip.mytestdomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 180.111.222.333:5060;rport;branch=z9hG4bKPjafb111a9-aaaf-4c9e-b456-748514855a06 From: <sip:555111222333@172.18.90.1 >;tag=88317259-052b-4997-bd2e-9c8836e27b47 To: <sip:555111222333@sip.netsip.net.au> Contact: <sip:9d43c8df-6378-4266-89c1-0a0bb0e1d0d1@180.111.222.333:5060> Call-ID: b7da52e2-dfd1-434e-a2c2-b28e1683ce0e -- Sincerely Jay

2 1

Fwd: Kamailio start failed
by Gopalakrishnan N 03 Jun '15

03 Jun '15

Hi, I have added the ASTERISK integration in kamailio.cfg, after adding those, am not able to start kamailio service, its getting failed, attached is the script. Any help would be appreciated. Thank you with regards, Gopalkrishnan N. Mob: +91 99404 91346 VoIP call - sip:saigop@gtalk2voip.com

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Need help to configure rtpporyx
by Mir Jee 03 Jun '15

03 Jun '15

Hi! Please help me to configure rtpproxy on centos + kamailio.

2 1

lookup() to fork execution of routing script?
by Andrey Utkin 03 Jun '15

03 Jun '15

Hi! Currently we have the following logic in our Kamailio routing script. On incoming INVITE, we lookup() the location of called party, then we analyse the updated Request-URI ($ru) regarding callee contact protocol: is it websocket-based or not, to distinguish web-based useragents from classic SIP agents and provide RTP profiles interconnection with rtpengine (RTP/SAVPF for web-based, RTP/AVPF for the rest). After that, we relay the mangled INVITE to callee location(s) with t_relay(). The issue arises when callee has both websocket-based and classic useragents registered simultaneously. According to lookup() manual, it replaces Req-URI with just one contact, and somehow also attach metainformation about additional locations, so that t_relay() forks the call. lookup() does not provide the possibility to fork the execution of routing script, so that we could execute our routing logic described above. But I would love this to be possible, because the only alternative way for calling to multiple locations seems to be the serial forking described in TM module manual. But we don't need serial calling contact-by-contact, waiting for call failures, we need parallel calling to all available user locations. So we need an advice how to go. Is there a way to implement what we need at scripting level, or should this feature be implemented in Kamailio code (or is it already there?). Thanks in advance for any help. -- Andrey Utkin

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Kamailio LCR Multiple choices
by Ali Taher 03 Jun '15

03 Jun '15

Hello, I'm using LCR module to send multiple choices. Everything works fine so far , yet I have a problem that last gateway comes first, then gateways are ordered correctly. Below is correct sorting of gateways: First gateway : 192.168.26.4 Second gateway: 192.168.26.1 Third gateway : 192.168.26.6 Below is Invite response I'm getting, as you can see 192.168.26.6 (last gateway) is included as the first gateway , then 192.168.26.4 (first gateway) comes next. Via: SIP/2.0/UDP 192.168.26.3:37066;branch=z9hG4bK-d8754z-885c37410729cc3b-1---d8754z-;rport= 37066;received=192.168.26.3 To: "9612"<sip:9612@192.168.110.181>;tag=b27e1a1d33761e85846fc98f5f3a7e58.8d8f From: "5000"<sip:5000@192.168.110.181>;tag=aa37be29 CSeq: 1 INVITE Call-ID: NTY0YjVmYzZiNzc3ZDA3ZDA3MmM0OGM1ZDBhZGZhNTM. Contact: <sip:9612@192.168.26.6>, <sip:9612@192.168.26.4>, <sip:9612@192.168.26.1>, <sip:9612@192.168.26.6> Below is routing block I'm using : if (is_method("INVITE")) { if (!load_gws(1, $rU, $fu)) { sl_send_reply("502", "Unable to load Gateways"); exit; } else while(next_gw()) { append_branch(); } sl_send_reply("300","Multiple Choices"); exit; } Appreciate your help. Regards, Ali

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delay on "location" table update
by Al S 03 Jun '15

03 Jun '15

Hi, I was wondering if it's normal for "location" table to be updated almost 10 sec after receiving REGISTER. Thanks,Al

2 1

Potential CNXCC bug
by Jay Patel 02 Jun '15

02 Jun '15

I am trying to use cnxcc for credit control . It seems like dialog and cnxcc modules are getting out of sync. For some reason , I see cnxcc.active_clients gets stuck and they never clear up. Interestingly only calls that gets failure 5xx and 4xx from PSTN gateway gets stuck, all successful calls are cleared . I have made sure , I have setflag(DLG_FLAG) 1st line in my routing logic. If you need my routing script, I can share with you . Am I missing anything obvious? As you can see , "dlg.list" is empty, ( no calls in progress in system.) kamcmd> dlg.list kamcmd> But , two user are stuck ( 1 with 6 calls in progress and another with 2 calls in progress) kamcmd> cnxcc.active_clients client_id:917044056840,number_of_calls:6,concurrent_calls:0,type:1,max_amount:0.000000,consumed_amount:0.000000; client_id:971554578191,number_of_calls:2,concurrent_calls:0,type:1,max_amount:0.000000,consumed_amount:0.000000; Here are the call dumps for both users. kamcmd> cnxcc.check_client s:971554578191 id:0,confirmed:no,local_consumed_amount:0.588000,global_consumed_amount:0.000000,local_max_amount:7.690000,global_max_amount:0.00000 id:1,confirmed:no,local_consumed_amount:0.588000,global_consumed_amount:0.000000,local_max_amount:7.690000,global_max_amount:0.00000 kamcmd> kamcmd> kamcmd> cnxcc.check_client s:917044056840 id:0,confirmed:no,local_consumed_amount:0.588000,global_consumed_amount:0.000000,local_max_amount:9.990000,global_max_amount:0.00000 id:1,confirmed:no,local_consumed_amount:14.100000,global_consumed_amount:0.000000,local_max_amount:9.990000,global_max_amount:0.0000 id:2,confirmed:no,local_consumed_amount:14.100000,global_consumed_amount:0.000000,local_max_amount:9.990000,global_max_amount:0.0000 id:3,confirmed:no,local_consumed_amount:14.100000,global_consumed_amount:0.000000,local_max_amount:9.990000,global_max_amount:0.0000 id:4,confirmed:no,local_consumed_amount:14.100000,global_consumed_amount:0.000000,local_max_amount:9.990000,global_max_amount:0.0000 id:5,confirmed:no,local_consumed_amount:0.588000,global_consumed_amount:0.000000,local_max_amount:9.990000,global_max_amount:0.00000 kamcmd>

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Kamailio LCR rule matching
by Ali Taher 02 Jun '15

02 Jun '15

Hi, I'm using LCR module in Kamailio , where matching of the rule (From-URI in LCR-rule table) is currently based on user part of sip address in From field (highlighted below in blue). While we need to change the identification to be on the otg value sent in the sip address in From field (highlighted below in yellow). INVITE sip:86532917453592@172.16.5.108:5060 <http://sip:86532917453592@172.16.5.108:5060> SIP/2.0 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 Record-Route: <sip:sansay9906915rdb2243@172.16.5.112:5060;lr;transport=udp> To: <sip:86532917453592@69.41.186.186:5060 <http://sip:86532917453592@69.41.186.186:5060> > From: <sip: <http://sip:+249964642906@216.53.4.5:5060> +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:+249964642906@216.53.4.5:5060 <http://sip:+249964642906@216.53.4.5:5060> > P-Asserted-Identity: <sip:+249964642906@216.53.4.5:5060 <http://sip:+249964642906@216.53.4.5:5060> > Privacy: none User-Agent: VCS 5.9.2.42-01 Allow: INVITE, ACK, CANCEL, BYE Max-Forwards: 69 Content-Length: 0 How can we achieve this ? Any help will be appreciated. Regards, Ali

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RTPengine+Kamailio, 200 OK without DTLS fingerprint
by Vasiliy Ganchev 02 Jun '15

02 Jun '15

Hello list! Have some difficulties with RTPengine: My test scheme is: UA(WS) -> Kamailio (WS proxy) -> Kamailio (SIP REGISTRAR, proxy) -> ASTERISK (media server) -> UA (SIP) Kamailio (WS) use RTPengine. Kamailio (WS proxy) - 4.2.3 Kamailio (SIP REGISTRAR, proxy) - 4.1.4 RTPengine - 3.3.0.0+0~mr3.8.0.0 I make a test call and have troubles in negotiation between UA(WS) -> Kamailio (WS proxy): UA(WS) -> INVITE -> Kamailio WS ...... rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force"); ...... this INVITE consists in SDP section a=fingerprint:sha-256...... When Kamailio (WS) receives 200 OK, it is also handled by RTPengine ....... rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force"); ....... but UA (WS) receives 200OK without fingerprint, and log an error: Failed to set remote answer sdp: Called with SDP without DTLS fingerprint. Currently spend a lot of time reading Kamailio/RTPengine documentation/mail-list - but without success If someone have some hints, I would appreciate any help. -- View this message in context: http://sip-router.1086192.n5.nabble.com/RTPengine-Kamailio-200-OK-without-D… Sent from the Users mailing list archive at Nabble.com.

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RFC 6140
by Al S 02 Jun '15

02 Jun '15

Hi, I was wondering if Kamailio support RFC 6140 to register Addresses of Record (AORs) in bulk map to a unique set of contacts. REGISTER sip:kamailio.example.com SIP/2.0 Via: SIP/2.0/UDP 198.51.100.3:5060;branch=z9hG4bKnashds7 Max-Forwards: 70 To: <sip:pbx@kamailio.example.com> From: <sip:pbx@kamailio.example.com>;tag=a23589 Call-ID: 843817637684230@998sdasdh09 CSeq: 1826 REGISTER Proxy-Require: gin Require: gin Supported: path Contact: <sip:198.51.100.3:5060;bnc> Expires: 7200 Content-Length: 0 Thanks,Al

2 3

next via branch value
by Juha Heinanen 01 Jun '15

01 Jun '15

when kamailio forwards invite request, it adds to it a new via header. for example, incoming invite: SIP/2.0/TCP 192.168.43.192:48089;branch=z9hG4bK4e1ab219cbde6190;rport outgoing invite: SIP/2.0/TCP 192.98.102.10;branch=z9hG4bK27b9.597e97c43da7ddd6be8520b662d616b9.0;i=5 SIP/2.0/TCP 192.168.43.192:48089;received=192.98.102.10;branch=z9hG4bK4e1ab219cbde6190;rport=49582 is it somehow possible to find out in the config script, what will the branch value of the new via header be? if that value would be known, it could be assigned to rtpengine extra_id_pv pseudo variable before making the offer and could then be used in rtpengine_delete to delete the right branch. if the new via branch value cannot be known, any suggestions on how to uniquely identify the outgoing branched of the invite? -- juha

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Request of advice about TAS for IMS
by dfretes＠ing.una.py 01 Jun '15

01 Jun '15

I am trying to setup a really simple (I hope so) IMS platform, with some basic services (like voicemail, IVR, videoconference), but I'm kind of stuck. I have installed Kamailio IMS on debian and a OpenIMS VM also. But my little problem begins when I try to configure a application server (in this case, a TAS). I have tried with Elastix 3.0 (kamailio+asterisk), asterisk alone, and read some about Restcomm. For Elastix and asterisk, I get stuck trying to setup the trigger point. For Restcomm, I only find documentation for integration with Clearwater IMS. So, I need your help. Somebody has something like that working? What TAS (or components for a TAS) do you recommend? Tutorials, examples, manuals that you know can help me? Thanks for reading, and many many thanks for any advice. David

2 1

Select a network
by Nelson Migliaro 01 Jun '15

01 Jun '15

Hello everybody, I need to apply a specific configuration to softphones registering from a specific network range. Is there something like:

2 2

Invalid SDP line (no '=' delimiter) error !
by Badri Ranganathan 01 Jun '15

01 Jun '15

Hi all, I am answering a SIP call made to my softphone. The softphone rings and also sends a 200 OK correctly. But the other end that made the call is seeing a failure in the SDP of the 200 OK packet. This error is seen in SDP header => Invalid SDP line (no '=' delimiter) error w.r.t w_frame4_public_ip.pcap It looks like the failure as seen in wireshark is because line "audio 5000 RTP/AVP 8 101" should really be "m= audio 5000 RTP/AVP 8 101" so it doesn't see the '=' delimiter. But if you look at w_frame52_local_ip.pcap, the packets sent from the local IP has the contents correct. It is only after passing the public IP and at the network premises, that this failure is reported by the Session Border controller. I am not able to understand why a correctly formed packet sent out is seen "corrupted" at the network end. Any ideas really appreciated. Thanks, Badri.

2 1

Re: [SR-Users] Regarding presence conference
by Daniel-Constantin Mierla 01 Jun '15

01 Jun '15

Hello, the conference mixer is supposed to send PUBLISH requests with information about who is in the conference room, who is speaking at that moment and so. That is not the job of the presence_conference module. Eventually, that could be a pua_conference module that has to be implemented. The presence_conference module was implemented as part of GSoC and the PUA was supposed to be inside SEMS, but the other project failed. If you want to get it all working, you have to implement the PUA either in your version of rtpproxy or part of kamailio as well. When all in kamailio, you can see the pairs of modules such as presence_dialoginfo and pua_dialoginfo. You already have presence_confgerece. Cheers, Daniel On 28/05/15 14:28, Surendra Pullaiah wrote: > > Hello Daniel, > > > > I am implementing conference bridge using kamalio and > RTP porxy without media mixing. Since for now we are ok with this. > My question is as per the flow of RFC 4579, UA may send SUBSCRIBE to > the conference URI with event as conference. In kamalio modules > presence_conference where they mentioned such way that it will support > the same what I am looking for. But I simulated using sipP by sending > subscriber with conference event. I am not seeing any notify XML which > will have the conference users list. One more doubt where it will pick > the conference users, since I am not seeing any specific data base > table for the same. > > > > Can you please clarify what exactly for > presence_conference module. > > > > Note: I have loaded three modules presence, presence_xml, > presence_conference, in route config I am calling handle_subscribe > when SUBCRIBE method receives. > > > > Regards > > Surendra > > India: +91-8124625001 > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 0

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

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"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

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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

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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

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