April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Building latest Git master branch with ZRTP enabled
by Jason White 27 Apr '10

27 Apr '10

Upon trying to build the latest code from Git with ZRTP enabled I get: src/switch_rtp.c: In function ‘read_rtcp_packet’: src/switch_rtp.c:2090: error: cast from pointer to integer of different size src/switch_rtp.c:2097: error: assignment makes pointer from integer without a cast src/switch_rtp.c:2105: error: ‘ret’ undeclared (first use in this function) src/switch_rtp.c:2105: error: (Each undeclared identifier is reported only once src/switch_rtp.c:2105: error: for each function it appears in.) src/switch_rtp.c:2106: error: label ‘end’ used but not defined src/switch_rtp.c:2102: error: label ‘do_continue’ used but not defined Debian x86_64 architecture, gcc 4.4.3. I haven't looked at the code in detail, but it is all enclosed in #ifdef preprocessor directives, so perhaps surrounding code has been changed without updating the ZRTP-dependent portions. Since I'm not familiar with the changes, the fixes weren't obvious. It's also possible that I tried to build at just the wrong moment.

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freeswitch, openzap, dahdi issues...
by Tom Christensen 27 Apr '10

27 Apr '10

In my quest to replace an unstable and deadlock prone asterisk solution I was able to get calls working over my PRI last night using freeswitch, native openzap, and dahdi 2.2.1. However, in order to make it work, I had to disable all echo cancellation in dahdi. If any of the 4 echo cancellation algo's were specified in /etc/dahdi/system.conf then there was no audio at all on the calls. Signaling worked, and calls could be made/received, but no audio in either direction. As soon as I disabled EC, audio worked (although, there was a good amount of echo). Also, I had intermittent (like 1 out of 2 or 1 out of 3) calls with very poor voice quality (robotic sounding, distortions, dropping in and out). This experience leads me to believe that possibly openzap isn't ready for prime time? Would I have been better served to include libpri in the technology stack? At this stage my "solution" looks like leaving asterisk where it is, and using it as a media gateway, passing calls off to freeswitch via SIP or IAX and using freeswitch as the app/registration/feature server. I don't know if that will necessarily be better, but I've had good luck with asterisk and call quality tuning (the PRI here has always had pretty bad echo, but I've been able to get asterisk EC working quite well on it), and it seems like asterisk has most of its issues with stability in transferring/parking/hold/voicemail/music on hold... I'm running a current FS SVN from 3 days ago, dahdi 2.2.1, and openzap using a digium te110p. The PRI is an NI2 t1 if that matters... Any ideas or recommendations of things to try would be greatly appreciated. I'd really like to avoid the dual server situation if at all possible. Thanks, -Tom

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g.729 Licenses from Freeswitch.org
by Ken Fulmer 26 Apr '10

26 Apr '10

1. Are these licenses additive? In other words, if I've purchased one for a machine and need a total of five, can I purchase four more and add them to the mix? 2. If we have to replace a machine, how can we replace the licenses without repurchasing them? Thanks, Ken Fulmer

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Any breaking change on register with 1.0.6
by Phillip Jones 26 Apr '10

26 Apr '10

Hi there, I have been using my Cisco phones registered with FreeSWITCH (1.0.4) for months now. Today I upgraded to 1.0.6 and they no longer registers. This what configured to get them to work. 1) Add <list name="lan" default="allow"> <node type="allow" cidr="71.**5.60.***/32"/> to acl.conf.xml 2) Add the appropriate directory entry. That worked (may be it shouldn't have done) - but on 1.0.6 (today's trunk) is fails. Is something else required in 1.0.6 do you know? Thanks Phil

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jBilling Setup on freeswitch
by Shrouk Khan 26 Apr '10

26 Apr '10

hi , has anyone been able to integrate freeswtich with jbilling ? swearching google did not come up with much answer , but it is claimed that freeswitch works with jbilling . I already tried astpp , but it wasnt very impressive. Can anyone please point me to some resources or tips on that ? :) -- Regards Shrouk Khan (Khan)

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some questions about cluster FS by using ODBC in the core.
by Nguyễn Mạnh Hùng 26 Apr '10

26 Apr '10

Hi list. I try to deploy cluster FS by using "ODBC in the core". But i have some problem when deploy: 1. I deploy 2 FS share registration ( by config odbc-dsn in internal profile). With default config: User 1001 registers in fs1: 10.2.48.194, user 1003 registers in fs2: 10.2.48.243. I show sofia status profile internal and see to registration: Call-ID: d3298837aa18ac2bOGYyNTYyZGJjNzcyNzNhNDA3MTI0MmY4MmMzOWFkZDU. User: 1001(a)10.2.48.194 Contact: "1001" <sip:1001@10.2.48.184:42064;rinstance=22b332e0ef7c7fc1> Agent: eyeBeam release 1003s stamp 31159 Status: Registered(UDP)(unknown) EXP(2010-04-16 11:26:47) Host: FS1.voip.vn IP: 10.2.48.184 Port: 42064 Auth-User: 1001 Auth-Realm: fs.vn MWI-Account: 1001(a)10.2.48.194 Call-ID: NzFiZmQxM2FlZDdmZjU3NTk4NjY4NmIwOWQ4NTcwZDI. User: 1003(a)10.2.48.243 Contact: "100311" <sip:1003@10.2.48.184:19374;rinstance=a7557f050b39a086> Agent: X-Lite release 1104o stamp 56125 Status: Registered(UDP)(unknown) EXP(2010-04-27 12:18:37) Host: localhost.localdomain IP: 10.2.48.184 Port: 19374 Auth-User: 1003 Auth-Realm: fs.vn MWI-Account: 1003(a)10.2.48.243 user 1001 can't dial user 1003 and vice-versa. It likes because user 1001 calls 1003(a)10.2.48.194 while user 1003 is registers 10.2.48.243. I read in list that can call between FS with odbc in the core. Do i miss config something ? 2. I have read some comments with 2 FS, we need to config DNS-SRV for load balance between them. But I wonder how gateway connects with fs by load-balance SIP trunk? Tks for help. Nguy�n Mạnh H��g Team Member - Telecom Security Bkis Telecom Dept. Bkis Internet Security Office: HH1 Building - Yen Hoa, Cau Giay, Hanoi Tel: (84 4) 37677090 ext 101 Mobile: (84) 1656 722 375 Email: HungNgM(a)bkav.com.vn Website: www.bkav.com.vn ____________________________________________ Do your best, the rest will come ! H繦 l� việc hết m・h, những điều tốt đẹp sẽ đến với bạn !

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Caller ID on inbound calls on Polycom
by Wellie Chao 26 Apr '10

26 Apr '10

I recently updated my FreeSWITCH installation from 1.0.4 to 1.0.6 and now I see a strange behavior with caller ID on Polycom phones when handling inbound calls. Here is the scenario: * Call is from 212-555-2222 (external number not on my softswitch) * Call is to 212-555-1001 (number on my softswitch, extension 1001) * extension 1001 is a Polycom phone (it's an IP301, but same problem occurs on other Polycom models such as the 501 and 601). SIP 3.1.4 and SIP 3.1.6 both affected. SIP 3.2.3 doesn't run on the IP301/501/601. * on phone 1001, the caller ID will display 212-555-2222 while ringing. The moment I pick up, the display will change to "From: 1001" (referring to the extension of the phone itself). Has anyone else experienced this problem, and does anyone know how to fix it? It does not occur with the snom 320 (and I assume it does not occur with any of the snom models based on extrapolation). While the problem only started when I updated from 1.0.4 to 1.0.6, it's entirely possible it's a configuration setting on the Polycom rather than a FreeSWITCH issue. I'm not sure. Anyone have pointers about how to resolve the problem (or even how to go about investigating the root cause)?

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Conferene under Xen
by Rudolf Denert 26 Apr '10

26 Apr '10

Hallo everybody, I have one question. Who does freeswitch generate the nessessary clock for real-time applications like conferencing, moh, ... The asterisk needs ztdummy. But there is a problem with running astersik on it. FS don´t have this problem on xen. Best regards

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How to handle and parse "detect speech" answer?
by patrick 26 Apr '10

26 Apr '10

detect speech described here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_detect_speech I have some recog result in freeswitch console: ................. Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false Content-Length: 165 <?xml version="1.0"?> <result grammar="holdr"> <interpretation grammar="yesno" confidence="98"> <input mode="speech">YES</input> </interpretation> </result> .................... They say: The XML body at the end there with our result has a Content-Length of 165. That is included as part of the overall count of 1791 so don't get tripped up parsing that. 1. How can I get this answer from console to any variable in my dialplan.? 2. How and when the best way to parse it? -- С уважением, Равальдини Патрик Стефанович Инженер по тестированию ООО «Центр речевых технологий» Тел.: (812) 325-8848, доб. 6225 Факс: (812) 327-9297 E-mail: patrick(a)speechpro.com http://www.speechpro.ru

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how i get configskypenew.tgz
by angels_ldc 26 Apr '10

26 Apr '10

I want install Skypopen cp /mnt/root/configskypenew.tgz ./ tar xzf configskypenew.tgz chown root.root .Skype But where are configskypenew.tgz ??

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