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Compile issues
by Tim King 19 Jul '10

19 Jul '10

I am trying to compile form the latest git and I keep getting stuck here. Any ideas? /bin/sh ./libtool --tag=CC --mode=link gcc -g -O2 -o pcregrep pcregrep.o libpcreposix.la gcc -g -O2 -o pcregrep pcregrep.o ./.libs/libpcreposix.a /usr/src/freeswitch/libs/pcre/.libs/libpcre.a g++ -DHAVE_CONFIG_H -I. -g -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See <file:///usr/share/doc/gcc-4.3/README.Bugs> for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/usr/src/freeswitch/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2

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DIsplay Name issue
by mike.burlingame 19 Jul '10

19 Jul '10

Hello, I need some help with the display name in FS here is my issue. Currently with FS if the A-LEG call invite is in the following format From: <sip:82617268@IPADDRESS>;tag=15074010004646399782 FS then will convert this to From: "82617268" <sip:82617268@IPADDRESS>;tag=82DmgpptX99ym What I need to be able to get FS to do is on invites that the display-name is blank that the output is also blank so no "" or no "NUMBER" based on the SIP RFC section 8.1.1.3 it states that the following methods are valid From: "Bob" <sips:bob@biloxi.com> ;tag=a48s From: sip:+12125551212@phone2net.com;tag=887s From: Anonymous <sip:c8oqz84zk7z@privacy.org>;tag=hyh8 So in the situation where there is no display-name presented to FS, FS should use option 2 in situations where FS does receive a display-name then option one is a valid choice. Thanks for your help in this matter..

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How to change Header (From value) while originate
by Durmuş Ali Öztürk 19 Jul '10

19 Jul '10

Hello, Using the *origination_caller_id_number *while doing a outbound call with the originate command has no effect of the caller Id. Because, our VoIP provider demands the caller ID into the From tag (From: <originator@domain>) of the SIP header, the *Remote-Party-ID* and * P-Asserted-Identity* will be ignored. Is there an alternative way to manipulate the From field value (which is taken from the username of a gateway profile by default)? Maybe another parameter in a originate command? Thank you for your answer. Ali

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Sipp auth with freeswitch
by Juan Antonio Ibañez Santorum 19 Jul '10

19 Jul '10

Hello! I am trying to do some load tests using Sipp with an authentication scenario. I can stablish one call using sipp as UAC but I receive continuously 407 SIP messages requiring authenticate. I've got one tcpdump capture: |Time | 192.168.2.230 | | | | 192.168.2.235 | |0,000 | INVITE SDP ( g711U) |SIP From: sip:sipp@192.168.2.230:5061 To:sip:9999@192.168.2.235:5060 | |(5061) ------------------> (5060) | |0,001 | 100 Trying| |SIP Status | |(5061) <------------------ (5060) | |0,021 | 407 Proxy Authentication Required |SIP Status | |(5061) <------------------ (5060) | |0,022 | ACK | |SIP Request | |(5061) ------------------> (5060) | |0,023 | INVITE SDP ( g711U) |SIP From: sip:sipp@192.168.2.230:5061 To:sip:9999@192.168.2.235:5060 | |(5061) ------------------> (5060) | |0,024 | 100 Trying| |SIP Status | |(5061) <------------------ (5060) | |0,190 | 200 OK SDP ( g711U telephone-event) |SIP Status | |(5061) <------------------ (5060) | |0,190 | ACK | |SIP Request | |(5061) ------------------> (5060) | |0,191 | RTP (g711A) |RTP Num packets:236 Duration:7.051s SSRC:0xDEE0EE8F | |(6000) ------------------> (2645) | |0,194 | RTP (g711U) |RTP Num packets:611 Duration:12.199s SSRC:0x5594757F | |(6000) <------------------ (26456) | |0,194 | RTP (g711U) |RTP Num packets:611 Duration:12.200s SSRC:0x5594757F | |(6000) ------------------> (26456) | |0,523 | 407 Proxy Authentication Required |SIP Status | |(5061) <------------------ (5060) | |0,525 | ACK | |SIP Request | |(5061) ------------------> (5060) | |1,524 | 407 Proxy Authentication Required |SIP Status | |(5061) <------------------ (5060) | |1,524 | ACK | |SIP Request | |(5061) ------------------> (5060) | |3,524 | 407 Proxy Authentication Required |SIP Status | |(5061) <------------------ (5060) | |3,524 | ACK | |SIP Request | |(5061) ------------------> (5060) | |7,524 | 407 Proxy Authentication Required |SIP Status | |(5061) <------------------ (5060) | |7,524 | ACK | |SIP Request | |(5061) ------------------> (5060) | |10,193 | BYE | |SIP Request | |(5061) ------------------> (5060) | |10,194 | 481 Call Does Not Exist |SIP Status | |(5061) <------------------ (5060) | |10,196 | BYE | |SIP Request | |(5061) ------------------> (5060) | |10,197 | 481 Call Does Not Exist |SIP Status | |(5061) <------------------ (5060) | |11,529 | 407 Proxy Authentication Required |SIP Status | |(5061) <------------------ (5060) | I respond with a new INVITE to first 407 received and RTP channels get connected but one and other 407 messages arrives during call and when I try to send BYE FS tells that call does not exists. 407 messages referes to first INVITE (Cseq=1). Any idea where the problem is? Regards

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How many active calls on the server
by leonardo alves 19 Jul '10

19 Jul '10

Hello, I want to get the information of how many active calls there are on my freeswitch server. I saw the mod_db that I can insert/remove the information I need from the db. Is this the right way to do it ? Or is there another one ? Thanks Leonardo

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ESL - playback
by Samuel Macedo 19 Jul '10

19 Jul '10

Hi, I have an application that use the ESL Outbound Connection to create an IVR Menu. Everything is running well, but I got a problem with the DTMF-Digit " *** ". I send a *playback* command with a file and wait for the DTMF to decide the action that will be taken. During this time - when I'm waiting - if the user sends a "*" DTMF-Digit, the FS stop the file automatically but I don't want this kind of action. Can I disable the end of files when I receive a * DTMF? Thanks, -- Samuel Macedo

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Sofia profile sip-ip parameter.
by Sergey Okhapkin 18 Jul '10

18 Jul '10

Is there a way to find from dialplan the value of sip-ip parameter of sofia profile by profile name? In some cases I need to initiate outgoing call to another SIP profile of the same FS instance, mod_loopback doesn't work properly to me because of CDR issues.

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Call For Assistance: Documenting FS and Icecast
by Michael Collins 18 Jul '10

18 Jul '10

FYI, I noticed that we don't have any information on the wiki on how to set up and use icecast2 with FS. Rupa mentioned in an earlier thread about using record or uuid_record to a shout://icecast_ip_addr:port/{uuid} syntax but I didn't get a chance to confirm the exact syntax. If you are using Icecast2 with FS please reply to this thread. I will be asking for your assistance with getting accurate how-to docs for using FS and Icecast. Thanks, Michael

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SIP Providers that SPAM
by Brian West 18 Jul '10

18 Jul '10

azdirectroute.com is a spammer... Anyone else get spam from them? /b

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Conference without members
by Samuel Macedo 18 Jul '10

18 Jul '10

Hi, I've experienced a strange behaviour in one of my conferences rooms. It's a conference with a lot of users and monitors - that can kick the users - witch was working fine. What is happening now is that anyone can hear anything in this conference. And when everybody leaves the conference, the conference still "alive" in the freeswitch - like above: freeswitch@internal> conference 1_conference xml_list <?xml version="1.0"?> <conferences> <conference name="1_conference" member-count="0" rate="8000" destruct="true" running="true" answered="true" enforce_min="true" dynamic="true"> <members></members> </conference> </conferences> There is no member in the conference, and when two members got into there none can hear. Do you have any idea about what is happening? Thanks, -- Samuel Macedo

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