January 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960
by Gary Chen 12 Jan '11

12 Jan '11

Sorry my mistake. It is Version 8.9 on Cisco 7960 ( POS3-08-9-00). Gary From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Peder Sent: Wednesday, January 12, 2011 11:12 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960 I’d say it is much more likely a Cisco 7960 bug. Cisco’s SIP software is total crap. It’s too bad because their phones are physically very nice and very reliable. I’d verify what release you are on as a search of Cisco’s site doesn’t show a 7.9. It shows up to 7.5. 8 goes all the way to 8.9. In * land, we never went past 7.4 as anything after that had issues. From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Gary Chen Sent: Wednesday, January 12, 2011 8:51 AM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960 FreeSWITCH Version 1.0.head (git-9350fb9 2010-12-07 00-20-07 -0600) I just install freeswitch and have two cisco 7960 phones (firmware sip 7.9) registered. I can call out to our service provider or make 7960 call each other. They all work fine except this: If I call into freeswitch echo test extension or check voicemail using one of Cisco 7960, the call will be droped after 30 seconds. By looking at SIP messages, it appears that After FS send out 200 OK to Cisco 7960, it never received ACK sip messages back and timeout. Here is the debug info in Cisco 7960: [11:56:30:41354610] SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.144:5060;branch=z9hG4bK2be0821e From: "Line1" <sip:1007@xxx.xxx.xxx.177>;tag=00082166efcb015d72e42433-5254ac8f To: <sip:9196@xxx.xxx.xxx.177;user=phone>;tag=p9H5Qy5r993mB Call-ID: 00082166-efcb0007-04fe7b08-3c126cb0(a)xxx.xxx.xxx.144 CSeq: 102 INVITE Contact: <sip:9196@xxx.xxx.xxx.177:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9350fb9 2010-12-07 00-20-07 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 253 Remote-Party-ID: "9196" <sip:9196@xxx.xxx.xxx.177>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1294739962 1294739963 IN IP4 xxx.xxx.xxx.177 s=FreeSWITCH c=IN IP4 xxx.xxx.xxx.177 t=0 0 m=audio 24992 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 [11:56:30:41354614] SIPTaskProcessSIPMessage: Line filter: Determining destination line... [11:56:30:41354615] sip_sm_determine_ccb: Matched to_tag [11:56:30:41354616] sip_sm_ccb_match_branch_cseq: Method index not found [11:56:30:41354616] SIPTaskProcessSIPMessage: Error: sip_sm_determine_ccb(): bad response. Dropping message. [11:56:34:41355009] SIPTaskProcessListEvent: cmd = 0x160200 [11:56:34:41355009] SIPProcessUDPMessage: recv UDP message from <xxx.xxx.xxx.177>:<50195>, length=<1185>, message= Looks like cisco 7960 having trouble to process 200 OK message from FS. Is this a possible FS bug? Does anybody know how to fix it? By searching on the WEB, I can see that Asterisk 1.6 also has the same problem in some revisions and been fixed. Gary

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Re: [Freeswitch-users] mod_db limit and group
by Madovsky 12 Jan '11

12 Jan '11

Just corrected wiki page from Insert a group entry: <action application="db" data="insert:groupname:sipurl"/> Delete a group entry: <action application="db" data="delete:groupname:sipurl"/> to Insert a group entry: <action application="group" data="insert:groupname:sipurl"/> Delete a group entry: <action application="group" data="delete:groupname:sipurl"/> ----- Original Message ----- From: Madovsky To: Meftah Tayeb Sent: Wednesday, January 12, 2011 10:31 AM Subject: Re: [Freeswitch-users] mod_db limit and group ha ok mod_db and limit seems to work together now.. ----- Original Message ----- From: Meftah Tayeb To: FreeSWITCH Users Help Cc: Madovsky Sent: Wednesday, January 12, 2011 3:36 AM Subject: Re: [Freeswitch-users] mod_db limit and group http://wiki.freeswitch.org/wiki/limit see default dialplan have ton of examples Le 12/01/2011 05:48, Madovsky a écrit : Is there any example of limit and group use ? I don't understand how to use it. Thanks _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Meftah Tayeb inum: +883510001288000 Phone: +13602276297 Fax: +12538020313

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Possible Freeswitch bug with Cisco 7960
by Gary Chen 12 Jan '11

12 Jan '11

FreeSWITCH Version 1.0.head (git-9350fb9 2010-12-07 00-20-07 -0600) I just install freeswitch and have two cisco 7960 phones (firmware sip 7.9) registered. I can call out to our service provider or make 7960 call each other. They all work fine except this: If I call into freeswitch echo test extension or check voicemail using one of Cisco 7960, the call will be droped after 30 seconds. By looking at SIP messages, it appears that After FS send out 200 OK to Cisco 7960, it never received ACK sip messages back and timeout. Here is the debug info in Cisco 7960: [11:56:30:41354610] SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.144:5060;branch=z9hG4bK2be0821e From: "Line1" <sip:1007@xxx.xxx.xxx.177>;tag=00082166efcb015d72e42433-5254ac8f To: <sip:9196@xxx.xxx.xxx.177;user=phone>;tag=p9H5Qy5r993mB Call-ID: 00082166-efcb0007-04fe7b08-3c126cb0(a)xxx.xxx.xxx.144 CSeq: 102 INVITE Contact: <sip:9196@xxx.xxx.xxx.177:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9350fb9 2010-12-07 00-20-07 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 253 Remote-Party-ID: "9196" <sip:9196@xxx.xxx.xxx.177>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1294739962 1294739963 IN IP4 xxx.xxx.xxx.177 s=FreeSWITCH c=IN IP4 xxx.xxx.xxx.177 t=0 0 m=audio 24992 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 [11:56:30:41354614] SIPTaskProcessSIPMessage: Line filter: Determining destination line... [11:56:30:41354615] sip_sm_determine_ccb: Matched to_tag [11:56:30:41354616] sip_sm_ccb_match_branch_cseq: Method index not found [11:56:30:41354616] SIPTaskProcessSIPMessage: Error: sip_sm_determine_ccb(): bad response. Dropping message. [11:56:34:41355009] SIPTaskProcessListEvent: cmd = 0x160200 [11:56:34:41355009] SIPProcessUDPMessage: recv UDP message from <xxx.xxx.xxx.177>:<50195>, length=<1185>, message= Looks like cisco 7960 having trouble to process 200 OK message from FS. Is this a possible FS bug? Does anybody know how to fix it? By searching on the WEB, I can see that Asterisk 1.6 also has the same problem in some revisions and been fixed. Gary

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Freeswitch eLearning
by Deon Vermeulen 12 Jan '11

12 Jan '11

Hi Does anyone know of an online training course for Freeswitch? Thank you very much Kind Regards Deon

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spandsp t38
by babak yakhchali 12 Jan '11

12 Jan '11

Hi I'm trying to send a fax through a cisco gateway like : fs -> cisco gateway -> pstn -> fax machine faxes are being sent fine (with some errors ) but there is a reinvite codec error (seems t38 fails): http://pastebin.freeswitch.org/14996 and this is the command I'm executing originate {origination_uuid=...,fax_ident=...,fax_header=...,fax_enable_t38=true,fax_enable_t38_request=true}sofia/external/...@(cisco ip) &txfax(...) shoud I do anything else to use t38? thanx

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mod_db limit and group
by Madovsky 12 Jan '11

12 Jan '11

Is there any example of limit and group use ? I don't understand how to use it. Thanks

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QoS in FreeSwitch
by Sergii Iefimov 12 Jan '11

12 Jan '11

*What kind of the QoS (Quality of Service)**do you have in FreeSwitch (DiffServ, IntServ or other**)?* * Does FreeSwitch offer the ability to mark the voice data with the proper tags so that our switch can prioritize the data through our network, or if it can't what hard ware device would you suggest to do this? Thanks! * Best Regards, *Sergii Iefimov* Mobile phone: +380 (68) 361 4627 E-mail: *efimserg(a)gmail.com*

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Profile configuration
by Kamigishi Rei 12 Jan '11

12 Jan '11

Hello, I have the following question regarding FS profile setup: is it considered normal to only have one profile at all? (The server is not behind a NAT/firewall; LAN users have direct routes to it.) From what I gather from the sample configuration, it's pretty much possible to distinguish between authenticated ("our own") users and external SIP requests via the dialplan. Quoting, <extension name="check_auth" continue="true"> <condition field="${sip_authorized}" expression="^true$" break="never"> <anti-action application="respond" data="407"/> </condition> </extension> <extension name="transfer_to_default"> <condition> <action application="transfer" data="${destination_number} XML default"/> </condition> </extension> Basically, I'd like FS to accept all calls (internal and external alike) on 5060, and use 5070 for external providers we register with. To get the "internal" profile to work like that on 5060, I have to comment the "apply-inbound-acl" setting (sip_profiles/internal.xml), and disable auth checking (internal_auth_calls=false in vars.xml). Would that be the correct solution, or is that somehow considered "insecure"? According to internal.xml, all non-authenticated (without user_context defined) calls fall into the public context anyway, and having an "is the user authenticated?" check in the public context allows us to transfer the call to the correct dialplan—so what are the security risks then? Speaking of which, is there a point in having user_context defined for users who need the default context, if they can just be redirected to that context via check_auth extension of the public context? Thanks in advance. -- Kamigishi Rei KREI-RIPE

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console
by Sam 12 Jan '11

12 Jan '11

A query, I have 2 FS running on one server on 2 different ips, so when i do fs_cli going to respective bins , i see console of only the first server. Is there any way to get the console of both the FS on the same server . I tried changing the port of event socket to 8022 but it donot works. <param name="listen-port" value="8022"/> Is there some method to start the console of both the instances. Regds Sam

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conference pin
by deniro 12 Jan '11

12 Jan '11

Hi using conferencing software and with the phone dialing, entering pin number it will go to a conference identified by pin in its default format it is "conference@profile+pin" in my case it will be "pin@profile+pin" since conference=pin. how do I do that? how do I provide a pin that takes me to conference which is identified by pin? thx deniro--

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