September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem with gateway registration
by Nicolas Brenner 08 Oct '09

08 Oct '09

Hello everyone, I am trying to add a gateway, but after configuring it just like the others gateways I have, it is failing to register with a message like this: 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed with status Operation has no matching challenge [904]. failure #1 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed Registration, setting retry to 10 seconds. I captured the sip traffic and noticed that when trying to register with one gateway (the one that works), I get a "Trying" reply immediately followed by a "401 Unauthorized" which contains a "WWW-Authenticate: digest" with a "qop=auth" parameter. Then Freeswitch replies with a second REGISTER including a large "Authorization: digest" section with cnonce and nc=00000001 parameters. The gateway which doesn't register, doesn't send the "qop=auth" parameter together with the "401 Unauthorized", and then Freeswitch sends a "Authorization: digest" section on the second REGISTER with no cnonce or nc parameters. I know very little abouth SIP, so I'm wondering what this "qop=auth" parameter means and how does it affect the registration process. Is there any way to do without the qop=auth parameter? Also, I tried registering with X-Lite directly to the gateway, and it worked, so it appears to be a problem in the Freeswitch/gateway combination. (Note: X-Lite sends an "Authorization: digest" section on the _first_ REGISTER, apparently this makes a difference) Attached is a sip trace for the registration traffic when doing "sofia profile external restart reloadxml" on the cli, captured with "tshark -i eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/capture.pcap -b filesize:51200 -b files:100 -R 'sip or rtp or icmp or dns or rtcp or t38'" Thanks! Nicolas

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Outbound INVITE rejected with 480 Temp Unavail, Reason "MANDATORY_IE_MISSING"
by Jerry Richards 04 Oct '09

04 Oct '09

Hello All, I have an internal extension that needs to send an INVITE without SDP body (Content Length 0). Freeswitch is replying with 480 Temporarily Unavailable with reason "MANDATORY_IE_MISSING". Would anyone know what I need to do to enable this? Best Regards, Jerry

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Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
by Mike van Lammeren 02 Oct '09

02 Oct '09

On Mon, Sep 28, 2009 at 9:05 PM, "Even André Fiskvik" <grevenx(a)me.com > wrote: > From: "Even André Fiskvik" <grevenx(a)me.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Mon, 28 Sep 2009 22:52:13 +0200 > Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey > I have been working with a similar setup myself, but for some reason I > ended up ditching theUltraMonkey setup because I just couldn't get it to > work right. > > It's been quite a while since my effort, so I don't remember what the exact > issue was. > I got registrations to work, but had some other sip-dialog issues. > > We have since then changed over to running OpenSIPs as a loadbalancer in > front of > multiple FreeSWITCH instances. This setup is still in testing, but > seemlingy works fine > (and if it doesn't, it's my own fault for writing a bad opensips config). > > After we have done some more testing I can create a wiki-page with config > details. > > > Best regards, > Even André > > Thanks, Even, that would be great! I might have to give up on the ultramonkey solution, since I can't find anyone who has made it work. It's too bad, because it would fit well with the rest of our architecture. Mike van Lammeren

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conference participant from behind NAT
by RobertT 02 Oct '09

02 Oct '09

I am a bit confused with what's going on in a following scenario. I have a public FS server with a public conference, that clients are connecting to with my softphone. All of this softphones have STUN option enabled and working, effectively resolving client's public IP address. They also have ICE enabled (but I guess it's not relevant here, since FS doesn't do ICE). Also, media trafic is secured with SRTP. The problem is when one client connects from port-restricted NAT into a conference he can hear sound for some time and he can be heard by other participants, but after awhile sound is gone and neither he hear anything nor he can be heard. Where is the problem? Is it NAT, closing RTP port after some silence period from client? I tried to start conference with waste flag, but without success eventually. The very same person can be contacted through this FS with direct call (being established in proxy_media mode) without any problems, but this is where ICE stuff starts doing its' "magic", I guess. Maybe I should try the same with SRTP disabled? Any help would be apreciated! Best regards, Robert.

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REGISTER fails with 407 after minutes of success register
by Fernando Testa 01 Oct '09

01 Oct '09

Hi all, I have a FS that registers on an Ericsson pabx as gateway under sip_external. This gateway start registering on the Ericsson ok, but after a while, around 50mins, it fails with the logs below. If I hit *sofia profile external restart* on fs_cli then the gateway returns to register with success (that means, we get 200 OK from Ericsson). This happens with FS 1.0.4 release tarball, and trunk r15011. I found similar situations on these links, but not actually found a solution. Any help is very welcome. *OS* CentOS 5.3 x86_64 4Gb RAM log at http://pastebin.freeswitch.org/10517 conf/sip_profiles/external/ericsson.xml: <gateway name="ericsson_1064"> <param name="username" value="1064"/> <param name="realm" value="10.227.0.3"/> <param name="password" value="5080"/> <param name="register" value="true"/> <param name="register-proxy" value="10.227.0.3"/> <param name="expire-seconds" value="60"/> <param name="retry-seconds" value="10"/> </gateway>

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Ringback when running G729 codec
by Hound Dog 30 Sep '09

30 Sep '09

Hi, very happy with freeswitch as a PBX/softswitch/SBC system its working solidly for a few weeks now - just great I have a question regarding ringback tones - custom or regular - I cant get freeswitch to send ringback using G729 I used the following settings ( it will just play one of the IVR prompts as ringback (filename ivr-to_repeat_these_options) - I took it from the G729 encoded files package , it has PCMA , G729 G723 extensions ) <extension name="inbound_routing"> <condition field="destination_number" expression="^4420885767(0\d)$"> <action application="set" data="ringback=/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options"/> <action application="set" data="instant_ringback=true"/> <action application="bridge" data="user/10$1"/> </condition> </extension> when I call with G711 enabled , it plays the file no problems - see log 2009-09-25 11:29:58.641361 [DEBUG] sofia.c:3289 Channel sofia/external/442078562107(a)80.80.80.80 entering state [early][183] 2009-09-25 11:29:58.641361 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562107(a)80.80.80.80 [BREAK] 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1745 Play Ringback File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 2009-09-25 11:29:58.641361 [INFO] mod_native_file.c:82 Opening File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 8000hz 2009-09-25 11:29:58.889369 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1006@82.80.131.233:40505 entering state [proceeding][180] when I call with G729 only - I get silence , and freeswitch only send the comfort noise packet and no RTP , see log 2009-09-25 11:28:57.437537 [DEBUG] sofia.c:3289 Channel sofia/external/442078562107(a)80.80.80.80 entering state [early][183] 2009-09-25 11:28:57.437537 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562107(a)80.80.80.80 [BREAK] 2009-09-25 11:28:57.685536 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1006@82.80.131.233:40505 entering state [proceeding][180] 2009-09-25 11:28:57.685536 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:1006@82.80.131.233:40505! mod_native_file works well for me when used in applications and plays G729 files no problem any ideas why is that happening , any suggestions on how to resolve ? thanks Ori

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Re: [Freeswitch-users] Need to know
by Ahmed Munir 30 Sep '09

30 Sep '09

Hi, Thanks for reply. Actually I'm currently running asterisk and for billing I'm using MySQL database. In the database i've listed various destination against their different rates. When call is made a script which queries MySQL for billing. My question is, how can I do this in FS using mod nibbles(note: I'm running asterisk in realtime)? And my other question is, will MySQL authenticates each time when call is made on FS? Kindly advise me. > ---------- Forwarded message ---------- > From: Ahmed Munir <ahmedmunir007(a)gmail.com> > To: FreeSwitch <freeswitch-users(a)lists.freeswitch.org> > Date: Wed, 30 Sep 2009 10:28:41 +0600 > Subject: [Freeswitch-users] Need to know > Hi, > > I want to use mod_nibble in freeswitch for billing purpose. I've got some > questions, as I'm listing down below; > > 1- How I create MySQL connection with FS? > 2- Is it possible to query the database when billing occurs? > 3- Will database credentials require each time when query to database? > 4- I'm currently running MySQL database in production for billing > purposes.Will it be okay to use my current database for mod_nibble? > > Kindly advise me soon and also please provide me the links as well. > > -- > Regards, > > Ahmed Munir > > > > > ---------- Forwarded message ---------- > From: William Suffill <william.suffill(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 30 Sep 2009 00:43:37 -0400 > Subject: Re: [Freeswitch-users] Need to know > mod_nibble uses ODBC. Via ODBC u can connect to mysql. > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core > > mod_nibble has it's own table design. > http://wiki.freeswitch.org/wiki/Mod_nibblebill > > It may be possible to tweak it to use your current tables but I'd > suggest getting the > basics working based on the documentation on the wiki and build from there. > > -- W > > > > -- Regards, Ahmed Munir

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Attention: behavior change with r15025
by Tamas 30 Sep '09

30 Sep '09

Hello, At r15025 the ringback tone handling has changed. The old method: When ringback was set, early media was ignored, meaning that if you got 183 with SDP, ringback was generated locally. The new way: When ringback is set, the ringback tone is generated only when there is no early media (thus 180 w/o SDP or 183 w/o SDP). When early media is presented, it will be used as ringing indication. To get the same behaviour as before r15025, you have to set the following: ignore_early_media=true ringback Some wiki pages were changed to report the new behavior. Many thanks for guys for fix/feature, especially for anthm and bkw. Kind regards, Tamas ps.: issue was related with FSCORE-454.

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Multiple Subnets
by Andrey Nepomnyaschih 30 Sep '09

30 Sep '09

Hello, I have some problems running a FreeSWITCH with endpoints located in different subnets. For example a FS is listening at 192.168.50.14/32 and endpoints from the same (192.168.50.0/24) subnet work as expected. But when I try to receive a media from an endpoint located at different subnet, let's say 192.168.60.0/24 the RTP stream cannot be bridged. FS sends an INVITE message asking endpoint to send media to external interface. Can anyone give me a hint where to look on how to overcome such behavior? Kind Regards, Andrey Nepomnyaschih

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FS Does Not Relay Presence PUBLISH To Subscribing Phones
by Jerry Richards 30 Sep '09

30 Sep '09

I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry

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