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FreeSWITCH overrides/dose no accept hangup cause
by ivdreg ivdreg 05 Oct '10

05 Oct '10

Hi All, I have 2 FS boxes that communicates each other. First originates call to second and on second FS perl scrips process a call. On some errors during processing I hangup call with specific hangup cause and everything is OK. For example: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 10.10.1.101;rport=5060;branch=z9hG4bK77698B1e1BZ3j From: "11111111111" <sip:11111111111@10.10.1.101<sip%3A11111111111(a)10.10.1.101> >;tag=X628H5HUDy86S To: <sip:999999999999@10.10.1.110:7060>;tag=Ztg1r5S60rQFF Call-ID: cca7076d-4b12-122e-44b6-7fd0651f7bfe CSeq: 2802191 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer *Reason: Q.850;cause=38;text="NETWORK_OUT_OF_ORDER"* Content-Length: 0 Remote-Party-ID: "999999999999" <sip:999999999999@10.10.1.110<sip%3A999999999999(a)10.10.1.110> >;party=calling;privacy=off;screen=no As you can see Q.850 cause is 38 - NETWORK OUT OF ORDER but on first FS application info says: variable_proto_specific_hangup_cause: [sip:503] variable_sip_hangup_phrase: [Service Unavailable] variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE] SIP Cause of both NORMAL_TEMPORARY_FAILURE and NETWORK_OUT_OF_ORDER is 503 and is the same but Q.850 cause is different. Is it normal behavior for 1.0.6 ? Is it also possible first FS to respect Q.850 cause ? Thanks

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Caller ID lost on second bridge in extension
by Mario 05 Oct '10

05 Oct '10

I have a 15 second ring to local extensions, then 15 seconds to local + 2 cell phones. However, when the second bridge is executed the caller ID is replaced by 0000000000. I tried removing pieces below like the execute app but it made no difference. Not found on the wiki or web either. Anyone know where/why it's lost and how to keep it? Thanks. Mario <extension name="u1phone"> <condition field="destination_number" expression="^2920$">  <action application="set" data="ringback=${us-ring}"/>  <action application="execute_extension" data="2921 XML default"/>  <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>  <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/>  <action application="set" data="call_timeout=15"/> <action application="bridge" data="group/home@${domain_name}"/>  <action application="execute_extension" data="2922 XML default"/> <action application="playback" data="/usr/local/freeswitch/conf/mvsounds/we_trying_please_hold.wav"/>  <action application="set" data="ringback=$${hold_music}"/>  <action application="set" data="ignore_early_media=true"/>     <action application="bridge" data="<originate_timeout=15>group/home@${domain_name}:_:sofia/gateway/uid1/19161234567:_:sofia/gateway/uid1/19162223333"/>    <action application="answer"/> <action application="voicemail" data="default ${domain_name} 100"/> </condition> </extension>

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RTCP support
by Dave Redmore 05 Oct '10

05 Oct '10

Hi All, I've done some investigating into getting RTCP support working, but am finding a lot of conflicting information. The WIKI says RTCP can be configured on a per session bases or for an entire profile. I've added "<param name="rtcp-audio-interval-msec" value="1000"/> to my profile config and restarted FS, but am not seeing any RTCP packets in my captures. Is there something else that needs to be done? Thanks. Dave Redmore Spigot Networks, Inc.

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api_on_answer
by Bernhard Suttner 05 Oct '10

05 Oct '10

Hi, I use api_on_answer to call a lua script. Within the lua script I want access to session variables but the lua variable "session" does not exists. How could I get access to them? Thanks in advance. Best regards, Bernhard

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No autoanswer with Originate call
by Eli Hayun 05 Oct '10

05 Oct '10

Hi When I am dialing from A phone to B phone and I do : <action application="set" data="sip_auto_answer=true" /> <action application="set" data="sip_h_Call-Info=<sip: ${domain_name}>;answer-after=2" /> everything is OK When I try from lua script api:executeString("originate [originate string] AUTO_ANS xml default") and in the dialplan I put the set autoanswer=true it is not working. any help? Eli

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Are there mandatory extensions in any context?
by covici＠ccs.covici.com 05 Oct '10

05 Oct '10

Hi. I notice that there are a few extensions at the beginning of the default context in the dialplan such as unroll-loops and a few more. I wonder if I make a new context for some specific purpose, to hold conferences, for instance -- do I need such extensions in that context as well? thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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High Availability and hostname used by Sofia
by Leon de Rooij 05 Oct '10

05 Oct '10

Hello List, This is going to be a long mail, please bear with me :-) I'd like to configure High Availability for a FreeSWITCH installation by using a failover server. I already discussed it a bit with Sekil on IRC, but I thought it'd be useful to discuss it here as well. The idea is as follows: - I have 2 servers both with an identical FreeSWITCH installation, FS1 and FS2. - Both servers have different static IP's but also one floating IP by means of heartbeat / haresources. - Both FS instances have one Sofia SIP profile that use the floating IP. - Both FS instances are always started (so the backup server will fail to load its SIP profile because the IP is not configured - which is expected behaviour) - There'll be a /etc/ha.d/resource.d "resource script" that starts the SIP profile (through fs_cli -x) the moment it becomes primary - The resource script will also send a "sofia recover" after the SIP profile has been brought up This way: - There'll be one logical switch. - The moment FS2 becomes primary, users will be reachable through the entry in 'sip_registrations' table in the ODBC db, even though they may have registered on FS1 - FS2 will start to send SIP options to NAT'ed useragents (keepalive / udp hole punching) the moment the SIP profile is started. - No client useragent involvement is necessary when failing over (like re-register) - Running calls will be continued because of the "sofia recover" Only problem here is that when a useragent registers to us, then FS also stores its own hostname in "sip_registrations" table in the columns "hostname" and "orig_hostname" (from mod_sofia_globals.hostname). This column is also used in the SELECT query when bridging to a user - so that means if the user registered to FS1, then a failover happens, then if a call is bridged towards that user on FS2, it won't find it because it will search with FS2's hostname in the SELECT query, right ? If everything I'm writing so far is correct, then would it be alright to override the hostname in sofia.conf ? I think that would solve my problem. I already looked it up - it is first set in mod_sofia.c at line 4565 through gethostname(). It can be overriden in sofia.c (function config_sofia) around line 2845. It would probably also work to give both hosts an identical hostname in the OS, but I don't like that (because of rsyslog and more). I'm not convinced yet on how to proceed, please let me know if I'm going the wrong direction. If I'm successful on getting this running, I'll document everything on the wiki - also the haresources script I already wrote I'll put in my contrib dir (or elsewhere in main tree if preferred). Thanks for your time, Leon

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After several hours FS registers ITSP every minute, requires FS restart
by Mario 05 Oct '10

05 Oct '10

I have 2 accounts with an ITSP that with the SIP profile below. After Freeswtich has been up for several hours, especially during my overnight testing, the accounts start to register every minute. This would not be really bad except that incoming calls don't work and I have to restart FS. I have the sofia trace below that also includes 407 Proxy Authentication Required messages but that does not seem to affect anything, the account works. Using the git version from this past Saturday: FreeSWITCH Version 1.0.head (git-f288e3c 2010-10-01 23-35-51 -0400). Thanks, Mario <include> <gateway name="uuid1"> <param name="proxy" value="sip.itsp.com"/> <param name="username" value="uuid1"/> <param name="from-user" value="uuid1"/> <param name="password" value="password"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> </gateway> </include> 2010-10-04 16:36:09.758277 [NOTICE] sofia_reg.c:342 Registering uuid1 send 865 bytes to udp/[216.143.130.36]:5060 at 23:36:09.759509: ------------------------------------------------------------------------ REGISTER sip:sip.itsp.com SIP/2.0 Via: SIP/2.0/UDP 106.88.82.100:5080;rport;branch=z9hG4bKamKeDBQrZ9j3m Max-Forwards: 70 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp> Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768939 REGISTER Contact: <sip:gw+uuid1@106.88.82.100:5080;transport=udp;gw=uuid1> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f288e3c 2010-10-01 23-35-51 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Proxy-Authorization: Digest username="uuid1", realm="rnktel.com", nonce="4caa64a90001484bedc7fa432b7fba4e53832e2474cb5555", algorithm=MD5, uri="sip:sip.itsp.com", response="7290bc4ec3cd32798c74d9ebd62c247e" Content-Length: 0 ------------------------------------------------------------------------ recv 567 bytes from udp/[216.143.130.36]:5060 at 23:36:09.866684: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 106.88.82.100:5080;rport=5080;branch=z9hG4bKamKeDBQrZ9j3m;received=209.143.244.127 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp>;tag=9214955d68c027e4c99a14a397962871.075c Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768939 REGISTER Proxy-Authenticate: Digest realm="rnktel.com", nonce="4caa64e500015288f99aa6d49e6686bbe3e8d9d066275ce8", stale=true Server: OpenSIPS (1.6.3-notls (i386/freebsd)) Content-Length: 0 ------------------------------------------------------------------------ send 865 bytes to udp/[216.143.130.36]:5060 at 23:36:09.867061: ------------------------------------------------------------------------ REGISTER sip:sip.itsp.com SIP/2.0 Via: SIP/2.0/UDP 106.88.82.100:5080;rport;branch=z9hG4bKBXc7e67Uvj9Ng Max-Forwards: 70 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp> Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768940 REGISTER Contact: <sip:gw+uuid1@106.88.82.100:5080;transport=udp;gw=uuid1> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f288e3c 2010-10-01 23-35-51 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Proxy-Authorization: Digest username="uuid1", realm="rnktel.com", nonce="4caa64e500015288f99aa6d49e6686bbe3e8d9d066275ce8", algorithm=MD5, uri="sip:sip.itsp.com", response="ea063d8a376643e491e1c41ce09dcf6e" Content-Length: 0 ------------------------------------------------------------------------ recv 549 bytes from udp/[216.143.130.36]:5060 at 23:36:09.975863: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 106.88.82.100:5080;rport=5080;branch=z9hG4bKBXc7e67Uvj9Ng;received=209.143.244.127 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp>;tag=9214955d68c027e4c99a14a397962871.aa06 Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768940 REGISTER Contact: <sip:gw+uuid1@209.143.244.127:5080;transport=udp;gw=uuid1>;expires=60, <sip:uuid1@209.143.244.127:5061>;expires=279 Server: OpenSIPS (1.6.3-notls (i386/freebsd)) Content-Length: 0 ------------------------------------------------------------------------ 2010-10-04 16:37:09.575144 [NOTICE] sofia_reg.c:342 Registering uuid1 send 865 bytes to udp/[216.143.130.36]:5060 at 23:37:09.623804: ------------------------------------------------------------------------ REGISTER sip:sip.itsp.com SIP/2.0 Via: SIP/2.0/UDP 106.88.82.100:5080;rport;branch=z9hG4bKc65Zg1rZSUZ8B Max-Forwards: 70 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp> Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768941 REGISTER Contact: <sip:gw+uuid1@106.88.82.100:5080;transport=udp;gw=uuid1> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f288e3c 2010-10-01 23-35-51 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Proxy-Authorization: Digest username="uuid1", realm="rnktel.com", nonce="4caa64e500015288f99aa6d49e6686bbe3e8d9d066275ce8", algorithm=MD5, uri="sip:sip.itsp.com", response="ea063d8a376643e491e1c41ce09dcf6e" Content-Length: 0 ------------------------------------------------------------------------ recv 567 bytes from udp/[216.143.130.36]:5060 at 23:37:09.723821: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 106.88.82.100:5080;rport=5080;branch=z9hG4bKc65Zg1rZSUZ8B;received=209.143.244.127 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp>;tag=9214955d68c027e4c99a14a397962871.9bd6 Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768941 REGISTER Proxy-Authenticate: Digest realm="rnktel.com", nonce="4caa652100015c59a0f47e6209ee434ff2e43a548d62af15", stale=true Server: OpenSIPS (1.6.3-notls (i386/freebsd)) Content-Length: 0 ------------------------------------------------------------------------ send 865 bytes to udp/[216.143.130.36]:5060 at 23:37:09.724214: ------------------------------------------------------------------------ REGISTER sip:sip.itsp.com SIP/2.0 Via: SIP/2.0/UDP 106.88.82.100:5080;rport;branch=z9hG4bKDFZrjv92p4NUQ Max-Forwards: 70 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp> Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768942 REGISTER Contact: <sip:gw+uuid1@106.88.82.100:5080;transport=udp;gw=uuid1> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f288e3c 2010-10-01 23-35-51 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Proxy-Authorization: Digest username="uuid1", realm="rnktel.com", nonce="4caa652100015c59a0f47e6209ee434ff2e43a548d62af15", algorithm=MD5, uri="sip:sip.itsp.com", response="48be3b8135d84350c145baf028460cfb" Content-Length: 0 ------------------------------------------------------------------------ recv 549 bytes from udp/[216.143.130.36]:5060 at 23:37:09.824242: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 106.88.82.100:5080;rport=5080;branch=z9hG4bKDFZrjv92p4NUQ;received=209.143.244.127 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp>;tag=9214955d68c027e4c99a14a397962871.c855 Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768942 REGISTER Contact: <sip:gw+uuid1@209.143.244.127:5080;transport=udp;gw=uuid1>;expires=60, <sip:uuid1@209.143.244.127:5061>;expires=219 Server: OpenSIPS (1.6.3-notls (i386/freebsd)) Content-Length: 0

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Substitution of mod_g729 by mod_com_g729 causes core dump
by Jan Riedinger 04 Oct '10

04 Oct '10

I just tried to subsitute mod_g729 by mod_com_g729 by using an appropriate moduls.conf.xml. After a restart of Freeswitch, it crahses, when the first inbound call gets processed: freeswitch(a)sns-fs1.sns.eu> 2010-10-04 12:32:27.064152 [NOTICE] switch_channel.c:779 New Channel sofia/external/xyz(a)a.b.c.d [8e94789e-5a8d-4088-8ad1-cd5fb2be0c4f] freeswitch: src/switch_core_memory.c:448: switch_core_perform_alloc: Assertion `pool != ((void *)0)' failed. Aborted (core dumped) There seems to be no error, when mod_com_g729 gets loaded: 2010-10-04 12:32:21.856829 [INFO] mod_com_g729.c:243 Permitted G.729AB channels: 20 2010-10-04 12:32:21.856920 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 10ms 8000bps 2010-10-04 12:32:21.856925 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 20ms 8000bps 2010-10-04 12:32:21.856928 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 30ms 8000bps 2010-10-04 12:32:21.856933 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 40ms 8000bps 2010-10-04 12:32:21.856937 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 50ms 8000bps 2010-10-04 12:32:21.856943 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 60ms 8000bps 2010-10-04 12:32:21.856947 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 70ms 8000bps 2010-10-04 12:32:21.856974 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 80ms 8000bps 2010-10-04 12:32:21.856978 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 90ms 8000bps 2010-10-04 12:32:21.856983 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 100ms 8000bps 2010-10-04 12:32:21.856988 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 110ms 8000bps 2010-10-04 12:32:21.856993 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 120ms 8000bps 2010-10-04 12:32:21.857144 [NOTICE] switch_loadable_module.c:274 Adding API Function 'g729_count' 2010-10-04 12:32:21.857175 [NOTICE] switch_loadable_module.c:274 Adding API Function 'g729_used' 2010-10-04 12:32:21.857199 [NOTICE] switch_loadable_module.c:274 Adding API Function 'g729_available' 2010-10-04 12:32:21.857229 [NOTICE] switch_loadable_module.c:274 Adding API Function 'g729_info' I'm using FreeSWITCH Version 1.0.head (git-758f383 2010-09-30 13-30-36 -0700) If I use mod_g729 instead of mod_com_g729 Freeswitch is running stable. Does anybody have any ideas how to debug this issue? Thank you in advance Jan -- Jan Riedinger Phone : +49-30-39 73 19 66 Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64 E-Mail: riedinger(a)sns.eu SNS Consult GmbH ICQ : 163-237-041 Südwestkorso 49a MSN : jan(a)sns-consult.de 14197 Berlin GERMANY Skype : Jan Riedinger AG Charlottenburg - HRB 71973

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voicemail.tpl vs notify-voicemail.tpl
by Mario 04 Oct '10

04 Oct '10

I now have email working but could not find info about the differences between voicemail.tpl and notify-voicemail.tpl. They look the same and I wonder when each are used. I see they are referenced in voicemail.xml. Thanks. Mario

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