- FreeSWITCH-users - FreeSWITCH Mailing Lists

Video Passthrough issue
by THEKKEDATH, S (S) 08 Nov '10

08 Nov '10

Hi all I was trying to pass video through Freeswitch pbx and I am facing issue, my codec in H264 and I configured as mentioned the freeswitch documentation. But the problem what I am facing is when my video is coming to FS with H264 it is not passing the video to other participant in the bridge. I am able to see only black screen. Can any one tell me what is this issue? And how can I resolve it ? Thanks Sooraj Thekkedath Senior Software Engineer Alcatel-Lucent Ventures This email, and any attachments there to, are intended only for use by the addressee(s) named herein and contain Alcatel-Lucent confidential information. If you are not the intended recipient of this email, you are hereby notified that any dissemination, distribution or copying which amounts to misappropriation of this email and any attachments thereto, is strictly prohibited. If you have received this email in error, please immediately notify me and permanently delete the original and any copy of any email and any printout thereof.

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(no subject)
by Kane, Michael (mkane02) 08 Nov '10

08 Nov '10

Hi, I've got an issue with the way FreeSwitch is currently handling (un)Register requests. When my SIP client performs a registration to remove all its binding by using a Contact header with a value of "*", FS generates a corresponding 200 / OK response with a Contact header with a value of "*". My current SIP client does not like a Contact header being included the 200 / OK response in this particular scenario and flags a failure. Is there any way to configure FreeSwitch (or its embedded Sofia stack) to prevent it from adding a Contact header in this particular scenario? Thanks, Mike

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Dialplan with Intercall causes Error.
by Raja Rokkam 08 Nov '10

08 Nov '10

Hi, I get the below error when I try contacting another SIP service provider using a freeswitch dialplan: 2010-11-04 13:13:28.409290 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000(a)192.168.3.175 [afcd49a9-351c-4439-8f81-a4f4e45b5c99] 2010-11-04 13:13:28.415386 [INFO] mod_dialplan_xml.c:418 Processing 1000->5099 in context default 2010-11-04 13:13:28.418291 [NOTICE] switch_channel.c:669 New Channel sofia/external/+18662087191 [61b94ef6-bd16-4b27-9866-09e5694b28c2] 2010-11-04 13:13:28.547282 [INFO] sofia.c:662 Update Callee ID to "+18662087191" <+18662087191> 2010-11-04 13:13:28.547282 [WARNING] sofia_glue.c:3290 Hello, I see you have a Sonus! FYI, Sonus cannot follow the RFC on the proper way to send DTMF. Sadly, my creator had to spend several hours figuring this out so I thought you'd like to know that! Don't worry, DTMF will work but you may want to ask them to fix it...... 2010-11-04 13:13:28.553283 [NOTICE] sofia.c:4733 Channel [sofia/external/+18662087191] has been answered 2010-11-04 13:13:28.565282 [NOTICE] switch_ivr_originate.c:3103 Channel [sofia/internal/1000(a)192.168.3.175] has been answered 2010-11-04 13:13:28.610279 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode! 2010-11-04 13:13:28.610279 [ERR] switch_core_io.c:726 Codec G.729 decoder error! 2010-11-04 13:13:28.610279 [NOTICE] switch_ivr_bridge.c:637 Hangup sofia/external/+18662087191 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2010-11-04 13:13:28.631289 [NOTICE] switch_core_state_machine.c:185 sofia/internal/1000(a)192.168.3.175 has executed the last dialplan instruction, hanging up. 2010-11-04 13:13:28.631289 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1000(a)192.168.3.175 [CS_EXECUTE] [NORMAL_CLEARING] 2010-11-04 13:13:28.631289 [NOTICE] switch_core_session.c:1182 Session 2 (sofia/external/+18662087191) Ended 2010-11-04 13:13:28.631289 [NOTICE] switch_core_session.c:1184 Close Channel sofia/external/+18662087191 [CS_DESTROY] 2010-11-04 13:13:28.634279 [NOTICE] switch_core_session.c:1182 Session 1 (sofia/internal/1000(a)192.168.3.175) Ended 2010-11-04 13:13:28.634279 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/1000(a)192.168.3.175 [CS_DESTROY] Can you please let me know what could be the reason? I am using the latest Freeswitch server, and PJSIP for the softphone library. Thanks, Raja.

2 2

problem with google talk
by joy this 08 Nov '10

08 Nov '10

Dear all: I have a problem with google talk. I will be appreciated if someone can help me. “FreeSWITCH and Google Talk cannot connect to each other after a successful ringing and answering” (refer to “ http://wiki.freeswitch.org/wiki/Mod_dingaling”) I test it by: soft-phone (x-lite 3.0) --->FS (GT account1) --->GT account2 (GT software 1.0.0.105 or web page). Account2 rings and answers, but no sound can be heard. The GT account2 does not send any RTP packages out. The dial plan is as below: <extension name="gvoice_out"> <condition field="destination_number" expression="^gmail\+([^\@]+)\@?(.*)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="dingaling/thisjoy/$ 1(a)gmail.com"/> </condition> </extension> The jingle config is as below: <profile type="client"> <param name="name" value="thisjoy"/> <param name="login" value="thisjoy(a)gmail.com/gtalk"/> <param name="password" value="PWD"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="auto-login" value="true"/>  <param name="sasl" value="plain"/>  <param name="server" value="talk.google.com"/>  <param name="tls" value="true"/>  <param name="use-rtp-timer" value="true"/>  <param name="exten" value="1000"/>  <param name="vad" value="both"/>  <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> Although I tried several versions, it is still unworkable. Now I am not sure if it is the problem of GT version or FreeSwitch. Is there any link for downloading the actually latest version of google talk software? Sincerely yours, this joy.

1 0

registration fails after several hours - FS problem?
by Mario 07 Nov '10

07 Nov '10

I really need help on this as I have weeks into this problem. I thought I had it nailed but I guess not. After 5.5 hours I get: 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds. 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed Registration, setting retry to 30 seconds. 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed Registration, setting retry to 15 seconds. 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds. 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777 and no way to make/get calls until I restart FS. I did this: 1. log 7 2. sofia profile xxxx siptrace on for each profile/gateway 3. restarted router All three did not solve the problem. The trace and log produced no additional lines which is why I am wondering if FS has a problem since the trace shows no SIP activity. 3 gateways with 2 ITSPs 2 DSL/WAN lines, 1 static and 1 dynamic I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the external static ip. sofia status profile ... has the right ext ip nat_map status shows the dynamic (wrong) IP I tried starting with -nonat but that was worse the only way to fix is restart FS. I read the wiki on external nat, auto_nat and everything else many times. Thanks Mario

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ODBC and MSSQL
by Peter Hinman 07 Nov '10

07 Nov '10

We've got a MSSQL server that I was hoping to put to use as an ODBC data source for FS. I can connect via ODBS to the database from the FS box and run queries using tsql. FS also appears to be able to connect to the database, as it creates several tables when it starts up. After successfully creating tables, it appears to run a test query, receives an error from which it concludes that transactions are not supported and then disables ODBC. Is MSSQL expected to work with ODBC and FS? It does successfully create tables. Does this mean that FS disabling ODBC is a bug that should be reported in jira? Is there something I should be setting in the config other than <param name="core-db-dsn" value="DSN:USERID:PASSWORD" /> Thanks, Peter Log snippet below: 2010-10-05 15:37:33.169737 [INFO] switch_core_sqldb.c:1275 Opening DB 2010-10-05 15:37:33.234702 [ERR] switch_odbc.c:427 ERR: [begin;delete from channels where hostname='';delete from channels where hostname='';commit;] [STATE: 42000 CODE 102 ERROR: [FreeTDS][SQL Server]Incorrect syntax near ';'. ] 2010-10-05 15:37:33.234775 [ERR] switch_core_sqldb.c:404 SQL ERR [STATE: 42000 CODE 8180 ERROR: [FreeTDS][SQL Server]Statement(s) could not be prepared. ] begin;delete from channels where hostname='';delete from channels where hostname='';commit; 2010-10-05 15:37:33.234796 [ERR] switch_core_sqldb.c:1326 Transactions not supported on your DB, disabling ODBC 2010-10-05 15:37:33.235916 [INFO] switch_core_sqldb.c:1275 Opening DB Thanks, Peter

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Outgoing Gateway Configuration
by Darryl Barlow 06 Nov '10

06 Nov '10

I am just starting with Freeswitch and trialling a simple setup for my small business. At this stage I have setup 4 extensions, and have set up two sip provider accounts which work with both incoming and outgoing calls. I envisage eventually having 4 sip accounts amongst two providers (MyNetFone and PennyTel) and one pstn line on an ata, but one step at a time. Ideally, what I would like is for outbound calls to use the first available outgoing gateway. Is there some variable that I can use in a condition to test if a particular gateway is in use by another call before bridging to that gateway? An attempt to bridge to an outgoing gateway which is already in use by another call does not necessarily fail. One of my providers (Pennytel) allows multiple outgoing calls on the same line in some circumstances, but this has not been setup, and I'm not planning on using this functionality with Freeswitch. An attempt to dial out on this line when it is already in use results in a recorded message. As this subsequent call does not in fact fail it seems that the only way for me to skip this gateway and go to the next is to actually test if the first gateway is actually in use. Any suggestions would be much appreciated. regards, Darryl

2 1

Multimple aliases for one user id
by Boris Kovalenko 06 Nov '10

06 Nov '10

Hello! Is this possible to have multiple aliases (phone numbers) for one user id? For example, the user's primary id (and the phone number) is 1000 (internal number) and 73435230000 is an alias (external number). So may I use in directory something like: <user id="100" alias="73435230000"> </user> -- С уважением, Борис Коваленко ЗАО "Тагнет" (3435) 494991

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PinDr0p: A GaTec System to Trace Call Paths Across Multiple Networks
by mazilo 05 Nov '10

05 Nov '10

Someone recently posted this http://www.gatech.edu/newsroom/release.html?nid=61428 linky on a http://www.dslreports.com/forum/r25000639-General-PinDr0p DSLR VoIP Forum . The project is called PinDr0p and it uses Single-Ended Audio Features to Determine Call Provenance through a series of algorithms to detect and analyze call artifacts. From an engineering point of view, the algorithms employed perhaps use some sort of orthogonal transformations which may require a lot of CPU power to transform/analyze segments of audio streams in a time domain to a frequency domain. Nevertheless, I posted this information here to see if it may interest FS developers to incorporate and implement such a scheme into the next releases of FS if the source codes are made available publicly. After all, the project is funded by NSF. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/PinDr0p-A-GaTec-System-to-Tra… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

FS Not Sending 200 OK For ReINVITE
by Jerry Richards 05 Nov '10

05 Nov '10

Hello, I have a problem where FS is intermittently not sending a 200 OK to a re-INVITE request (it always sends 100 Trying). I posted FS logs of a GoodCall and the BadCall: Bad Call: http://pastebin.freeswitch.org/14422 Good Call: http://pastebin.freeswitch.org/14423 Can anyone see what is happening to the reINVITE? I can also provide Wireshark traces if that helps. Thanks, Jerry

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