September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Confusion about sip hangup cause Q850 hangup cause and long struggles with a provider.
by Sias Mey 10 Sep '12

10 Sep '12

Hi, Im sure this subject has been beaten to death .. but many googles and many email searches hasent really managed to find me something. Im a dev for a small company that writes call center software. Freeswitch was a godsend, thank you. Now .. the confusion. We are getting a lot of what seem to be strange hangup codes from a new provider big fights about loads of failled calls ensued blah blah.. much sip packet logging and manual inspection later.. I found the following. from xmlcdr. <sip_hangup_disposition>recv_refuse</sip_hangup_disposition> <sip_term_status>408</sip_term_status> <proto_specific_hangup_cause>sip%3A408</proto_specific_hangup_cause> <sip_term_cause>102</sip_term_cause> <hangup_cause>RECOVERY_ON_TIMER_EXPIRE</hangup_cause> <hangup_cause_q850>102</hangup_cause_q850> this just to show its the same call <sip_call_id>0d4d4e76-735a-1230-d2ac-000423b5571b</sip_call_id> and from the sip messages. SIP/2.0 408 Request Timeout Call-ID: 0d4d4e76-735a-1230-d2ac-000423b5571b Reason: Q.850;cause=18;text="no user responding" And according to the very useful wiki page on Q.850 codes 408 should = 18 like it does in the providers response. Why then is the q850 hangup cause in the CDR 102? and where does that translation come from. This is a single example but I also have loads and loads where the CDR claims q850 code 18 but the sip messages provide 31 or a range of other codes. I can understand if the q850 code from the sip message is not being read by FS since FS has to be a bit more agnostic than that and in the pas I have almost exclusively worked with direct connections to TDM hardware so my knowledge and understanding of the sip messages is rather limited. But even in that case, shouldent the q850 code in the cdr at least conform to the translation from the wiki page? Oh I am not currently running the latest git release, having some libtiff issues on ubuntu to compile. I will respond to this again if I manage that and it helps matters. Thank you for your time and help, Regards Sias

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simultaneous calls
by Zaid Tillma 10 Sep '12

10 Sep '12

On Sun, Sep 10, 2012 at 11:10 AM, Gabriel Gunderson <gabe at gundy.org> wrote: > What if they don't answer at the same time? Both of the extensions are set to auto-answer. In case one of them does not answer because it is being used, play on the non-busy extension. Once the busy extension gets un-busy, play the remaining audio on it (i.e does not have to start from the begining). I understand that this will lead to the case where one extension does not play the audio at all. Now in case both are busy, wait till one of them gets unbusy and then play. Then the plan listed above applies. FYI - I get a daily digest, and I am replying to Gabe's email by copy & paste from the digest. If this does not work, I'd appreciate someone letting me know how to post correctly if all you get is the daily digest. Thanks Z

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Odd SIP user registration problem
by Melvin King 10 Sep '12

10 Sep '12

Hello, I've been beating my brains out for a while with this problem, I think I might need some help. I've been trying to set it up as a trimmed down to the max carrier grade SIP PBX. Here's my setup I'm using a amazon ec2 64 bit image with a public static IP for the host. (Inside a NAT) ports opened from the outside are TCP 5060 UDP 16384 - 32768 FreeSWITCH Version 1.2.0-rc2+git~20120731T213556Z~e97da8e20a (1.2.0-rc2; git at commit e97da8e20a This is the relevant error message that I am getting, I cannot register any users at all, even though they're in the directory 2012-09-08 10:02:13.716511 [DEBUG] sofia_reg.c:1470 Send challenge for [ 1000(a)23.21.191.100] 2012-09-08 10:02:13.796513 [WARNING] sofia_reg.c:2442 Can't find user [ 1000(a)23.21.191.100] You must define a domain called '23.21.191.100' in your directory and add a user with the id="1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2012-09-08 10:02:13.796513 [DEBUG] sofia_reg.c:1414 Send forbidden for [ 1000(a)23.21.191.100] 2012-09-08 10:02:13.896514 [DEBUG] sofia_reg.c:1470 Send challenge for [ 1000(a)23.21.191.100] 2012-09-08 10:02:13.976510 [WARNING] sofia.c:5199 Ping succeeded flowroute with code 200 - count -1/1/1, state UP 2012-09-08 10:02:13.996519 [WARNING] sofia_reg.c:2442 Can't find user [ 1000(a)23.21.191.100] You must define a domain called '23.21.191.100' in your directory and add a user with the id="1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2012-09-08 10:02:13.996519 [DEBUG] sofia_reg.c:1414 Send forbidden for [ 1000(a)23.21.191.100] Any insight would be greatly appreciated Mel

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Enter and exit sound in a conference with wait-mod flag
by Emrah 10 Sep '12

10 Sep '12

Morning guys, I noticed something that sounds inconsistent and would appreciate if you could confirm it. I have a conference profile with the wait-mod flag set on. Upon entering the conference as a user, the MOH stops for current participants, the enter sound is played and MOH resumes. This is more or less a 1 second pause. However, when I exit the call as a user, the exit sound is played mixed with the MOH with no pause whatsoever. How can we manage the pause there? I would ideally want to be able to pause / resume the MOH in a conference in order to allow the enter / exit sound played along with the name of the caller, without the distraction of the MOH. I announce callers when they enter and exit the conference. Any good hints would be appreciated. Best, Emrah

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stereo recording problem - tracks out of sync
by Dave Horton 10 Sep '12

10 Sep '12

I have an application where I create a bridging conference that outdials a B party, and then I record the stream between the conference and the B party. However, the two channels of the resulting stereo recording are not in sync: for instance, if I call into an IVR and barge into a prompt by pressing a dtmf, on the recording the dtmf tone appears several seconds after the barged-in prompt stopped. Its as if the stream from the conference to the B leg has been shifted in time so it no longer matches up properly with the conference mix. I can't tell if this variance is constant or gets worse as the conference goes on. Has anyone a notion of what might be going on here? Here is my dialplan: <extension name="6174973895" continue="false"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="6174973895"> <action application="ring_ready"/> <action application="answer"/> <action application="start_dtmf"/> <action application="set" data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/> <action application="export" data="outfile=$${base_dir}/tmp/${call_id}.wav"/> <action application="lua" data="demo_call2.lua"/> <action application="export" data="RECORD_STEREO=true"/> <action application="export" data="nolocal:execute_on_media=record_session ${outfile}"/> <action application="export" data="nolocal:api_on_answer=conference ${call_id} play misc/this-call-may-be-recorded.wav async"/> <action application="conference" data="bridge: ${call_id}@simple:{origination_caller_id_number=${caller_id_number}}sofia/normal_customer/${outdial}@${egress_gateway}"/> </condition> </extension>

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Question about Exporting the cacaert.pem for client devices
by Andrew Carrega 10 Sep '12

10 Sep '12

I followed the Freeswitch wiki for enabling tls & srtp on Freeswitch. I have it enabled on my internal and external profiles and both profiles are starting up just fine. I can review my certificate details with the command: openssl x509 -noout -inform pem -text -in /usr/local/freeswitch/conf/ssl/agent.pem I am not at this section of the wiki where it says the clients should have at least the CA root certificate. Clients should all have at least the CA root certificate installed onto them in order to ensure security. Without enabling chain verification (that the server certificate was issued by the approved CA) a MITM attack is possible against a client. The CA certificate is the conf/ssl/cafile.pem it contains only a certificate and clients use it to ensure the server certificate is issued by the CA. Where I am stuck is understanding how to export or download the cacert.pem from the server? I seem to don't understand the process or tools to use and I can't seem to access /usr/local/freeswitch/conf/ssl directory or the /usr/local/freeswitch/conf/ssl/CA from root. Any help is appreciated.

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map inbound calls to a gateway based on IP
by Richard Zheng 10 Sep '12

10 Sep '12

Hi, I know that this has been discussed many times. Just want to get a clear answer. Is it possible to map inbound calls to a particular gateway based on IP address? Someone seems to claim that it's working. Others say no way. I tried to put context param in gateway section and no effect. This is in sip_profile gateway config. There were suggestions to just use context public, then send to its own context in dialplan. But that's not all the same. In asterisk world, the following section is sufficient in sip.conf. The system matches IP based SIP traffic first, then look for users with registration based. [providerA] type=peer qualify=yes context=provider_a host=1.2.3.4 disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference fromuser= dtmfmode=rfc2833 fromdomain=provider_a Richard

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effective_callee_id_name behaviour
by Yehavi Bourvine 09 Sep '12

09 Sep '12

Hello, Our FreeSwitch is connected to a Nortel PBX via E1-SIP gateway (AudioCodes). This gateway supports the P-Asserted-ID field. With a vanilla configuration the following happens when a FS user calls a user on the Nortel: - During the ringing phase the name of the callee is "outbound call" - When the other side answers the name of the callee is set to the one sent from the Nortel. In order to have the name also during the ringing phase, I set calle_id_name and effective_callee_id_name. After doing so, the name is not changed after the remote user answers. I need it to change to the name sent from the Nortel, as it shows the name of the one who actually answered the phone. I also tried adding ignore_display_updates=false, but it didn't change the behaviour. Any idea how to do it? Thanks! __Yehavi:

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record video to file of a two way call
by openser 08 Sep '12

08 Sep '12

Hi, I want to record video to a file of a bridged video call, i kown record_session app ,but it only record audio, and record_fsv app only record one call leg but not a two leg call, is there any app or what can i do for this ? thanks.

1 0

Best SIP phone?
by Emrah 08 Sep '12

08 Sep '12

Hi all, I have tried many IP phones and nothing comes close to the audio quality of a Polycom. I use a VVX1500 and everything is top notch, but it doesn't suit me entirely. 1. It's touch screen and I'm blind. 2. It's always a pain to restart the phone anytime you make a change in the provisioning configs. It's great in a corporate environment, but not so much for experimenting. 3. There isn't many ways to interface with the phone… E.g.: trigger a dial out from your computer's address book. I tried Snom a couple years ago and the audio quality was not good, with plenty of echo on speakerphone and buggy firmwares. Cisco is pretty much out of the question because of their closed configs and Linksys is pretty low end if my recollection is correct. What can you recommend that is stable, versatile, open and good quality? Cheers, Emrah

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