November 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

NoOp() equivalent?
by regs＠kinetix.gr 20 Nov '08

20 Nov '08

Is there an equivalent to asterisk's NoOp() so that I can print (within the XML dialplan) the stuff I want on the FS console? Like variables and stuff?

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Dialplan: how to check whether a User is registered?
by Peter P GMX 20 Nov '08

20 Nov '08

Is there a way to check if a user is registered locally? I have the following Scanrio: If a call comes in from an external gateway then Freeswitch1 shall check if the destination UA is registered locally. If not then redirect (302) the call to Freeswitch2. But how to determine that the UA who shall receive the call is registered here? Is there another way instead of - trying o bridge - and if it fails take the next condition? Best regards Peter

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Routing Calls
by paulo leonardo 20 Nov '08

20 Nov '08

Good morning list, i have a litle problem again ;-) ! 1 - I have a freeswitch (192.168.170.101) running on port (5060, 5061); - that's ok 2 - My freeswitch is registred to Proxy SIP (SER) (XXX.XXX.XXX.XXX); - that's ok 3 - I have a user local (1000) registred in my freeswitch; - that's ok 4 - When i make call (using user 1000) to another people registred in my SER, everything ok; <action application="bridge" data="sofia/gateway/softswitch/$1"/> 5 - I would like to recieve call in my freeswitch and redirect to my user 1000! But i get errors! *ERROR* 2008-11-20 10:58:29 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't find user [sipbxs00(a)sip.xxxxx.com.br] You must define a domain called 'sip.xxxxx.com.br' in your directory and add a user with the id="sipbxs00" attribute and you must configure your device to use the proper domain in it's authentication credentials. thanks

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Support for g729 and g723 codecs
by Pablo Hernan Saro 19 Nov '08

19 Nov '08

Hi there, Anyone knows if I can use Open G729-G723 from http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ with FreeSWITCH? I need transcoding capabilities. Another idea? I was googling for a while but I got nothing. Thanks in advance. Pablo

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Wrong IP on ACK?
by David Aldworth 19 Nov '08

19 Nov '08

For some reason when trunking with Asterisk PBX's (yes, I know) FS wants to send the ACK to the internal ip found in the Contact field of the 200 OK. We have the force rport setting on but it's still not responding to that IP. Register's work. Most of the sip signalling works, just when the customer specifies the Contact filed with an internal ip. Below is a packet capture and our external.xml conf file. U 2008/11/04 09:17:08.259672 64.74.188.23:5060 -> 68.188.189.202:5060 INVITE sip:989607XXXX@68.188.189.202:5060 SIP/2.0. Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bKc6y6pr80HyeaN. Max-Forwards: 68. From: "user" <sip:303452XXXX@64.74.188.23>;tag=Dt6v81cDZXa3B. To: <sip:989607XXXX@68.188.189.202:5060>. Call-ID: de7c471c-252e-122c-3cba-5f1bac93b621. CSeq: 106789378 INVITE. Contact: <sip:mod_sofia@64.74.188.23:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session- description, presence.winfo, message-summary. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 370. U 2008/11/04 09:21:32.861720 68.188.189.202:5060 -> 64.74.188.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 64.74.188.23 ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: <sip:989607XXXX@192.168.0.5>. Content-Length: 0. . U 2008/11/04 09:21:32.861845 68.188.189.202:5060 -> 64.74.188.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 64.74.188.23 ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>;tag=as1da4b7aa. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: <sip:989607XXXX@192.168.0.5>. Content-Type: application/sdp. Content-Length: 285. . v=0. o=root 10970 10970 IN IP4 192.168.0.5. s=session. c=IN IP4 192.168.0.5. t=0 0. m=audio 15876 RTP/AVP 18 0 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2008/11/04 09:21:32.862573 64.74.188.23:5060 -> 192.168.0.5:5060 ACK sip:989607XXXX@192.168.0.5 SIP/2.0. Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bK0r7c2a501SQvD. Max-Forwards: 70. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>;tag=as1da4b7aa. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 ACK. Contact: <sip:mod_sofia@64.74.188.23:5060>. Content-Length: 0. . external.xml <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings>

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FS is not suitable for MESSAGE, is it?
by Iñaki Baz Castillo 19 Nov '08

19 Nov '08

Hi, I'm starting with FS and still don't handle very well the dialplan syntax, but anyway I've realized that FS is not suitable for sending MESSAGE (instead of INVITE). I've registered two "internal" users (the default config): 1000 and 1001: a) INVITE between internal users: - 1000 calls 1001 - FS asks for authentication - 1000 resends the INVITE with authorization headers. - FS calls 1001 - 1001 answers - FS brigdes the call by sending 200 to 1000. b) INVITE to external user: - 1000 calls sip:bob@external_domain.com - FS asks for authentication - 1000 resends the INVITE with authorization headers. - FS replies 404. c) MESSAGE between internal users: - 1000 sends MESSAGE to 1001 - FS ***doesn't** ask for authentication <---- !!!!!!! - FS replies 200 OK directly - FS sends the MESSAGE to 1001 - 1001 replies 200 OK to FS d) MESSAGE to external user: - 1000 sends MESSAGE to sip:bob@external_domain.com - FS ***doesn't** ask for authentication <---- !!!!!!! - FS replies 200 OK directly - FS ***doesn't*** send the MESSAGE to the destination user. Well, I think it's obvious that FS MESSAGE handling is not coherent. Opinions? is there something to do? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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accountcode and user id not present in CDRs
by regs＠kinetix.gr 19 Nov '08

19 Nov '08

I am using acls (cidr) to accept incoming calls from a gateway that I do not want to register in my FS box. I have this gateway configured in a xml file : freeswitch/conf/directory/default/gateway1.xml <include> <user id="GATEWAY1" mailbox="" cidr="xxx.xxx.xxx.xxx/32"> <params> <param name="password" value="1234"/> </params> <variables> <variable name="accountcode" value="CUSTOMER1"/> <variable name="user_context" value="my_context"/> <variable name="effective_caller_id_name" value="gateway1_callid"/> <variable name="effective_caller_id_number" value="238383838383"/> </variables> </user> </include> I have the corresponding cidr in my ACL in acl.conf.xml. I am able to make a call from that gateway to my FS but in my CDRs (both xml or cdr_csv) the accountcode or user id is not present. Any help on how to define an endpoint (originating) and use some attribute (like account_code or user id) for billing purposes?

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Conference
by Baskar 19 Nov '08

19 Nov '08

Hi, I want to have the following scenario to be done on the conference room. 1. I am calling the 1st person and talking with him. 2. When I require, I need to hold the 1st person and dial out 2nd person. (till such time 1st person to be on hold by hearing hold music). 3. I need to unhold the 1st person so that three of us can be in the conference call. Kindly help me in getting this done. Thanks N.Baskar

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i need help regarding freeswitch
by Faisal Maqsoodi 19 Nov '08

19 Nov '08

Dear all, Hi I ve just started reading on freeswitch. >From where should i start? Basically, i am am a graduate electronic engineer. Can i be a successful developer of freeswitch? Its my first experience of any telephonic system. Faisal rehman, Pakistan

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Anthony is all heart ...
by Gonzalo Servat 19 Nov '08

19 Nov '08

Hi All, We all like to be thanked and when someone does something for me, I like to show my gratitude towards them and I think what Anthony did for me today deserves a public show of gratitude. It goes like this: I have a single port FXO card in my home machine running FS and I needed to modify the tone_detect app so that it would not incorrectly detect busy tones during phone calls. If I call in and hang up on the remote end, FS would detect the busy tone and hang up correctly. The problem was that tone_detect would act as soon as it detected the requested tone (in my case, busy tone) so I might be in the middle of a phone call it would sometimes incorrectly detect a single busy tone and drop the call. I tried to ignore this at first but it started happening more often so I spoke to Anthony about the possibility of modifying tone_detect to have a "hit count" during a time window, so that it would have to detect X number of busy tones before hanging up the call. As you can all see, this is probably a minor issue on Anthony's priority list since there are many more important issues to fix or things to do on FS than fix tone_detect. Even so, he spent a good 6 hours on this issue and kept hacking at it until he got it fixed. Anthony, once again, fixing tone_detect really helped me but what I appreciate more than anything is the willingness to help and invest so much of your time on this .. so thank you once again! - Gonzalo (aka znoG) PS: I will soon update the wiki page http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect with details on how to use the hit count

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