May 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Install without example configurations
by Lon Baker 10 May '09

10 May '09

Is there a way to compile and install without all the example configuration information? A truly clean install out of the gate. Lon

5 4

DTMF recognition flaky
by Jay Austad 09 May '09

09 May '09

Using the default installation, I've noticed that when I (or someone else) calls in on my SIP trunk and keys in an extension, not all of the numbers are recognized unless they hold the key down for at least 1/2 second to a second. Is there a way to improve DTMF recognition so people can just type in stuff without having to hold the keys down? -- jay austad | 612.423.1433 | austad(a)signal15.com

7 14

Gateway Outbound Dial Config Problem
by Dale 08 May '09

08 May '09

Hello, I am trying to get a new freeswitch installation working and I have it registering to a sip provider just fine and can receive inbound calls but when I try and place a call out through the gateway the switch is rejecting it due to the domain in the to of the INVITE. Here is the INVITE 15:15:19.467351 IP (tos 0x0, ttl 64, id 40045, offset 0, flags [none], proto: UDP (17), length: 1398) 207.4.223.35.5080 > 207.16.137.36.sip: SIP, length: 1370 INVITE sip:16173800299@207.16.137.36 SIP/2.0 Via: SIP/2.0/UDP 207.4.223.35:5080;rport;branch=z9hG4bKcc3a14Zcg8NZp Max-Forwards: 69 From: "FreeSWITCH" <sip: 6172384723(a)172.16.12.100;transport=udp>;tag=54FXray2SNaKN To: <sip:18023800299@207.16.137.36> Call-ID: 6956c6eb-b6a7-122c-5fb1-725bd701687a CSeq: 114786724 INVITE Here is the REGISTER for the same gateway 15:24:27.746984 IP (tos 0x0, ttl 64, id 40075, offset 0, flags [none], proto: UDP (17), length: 661) 207.4.223.35.5080 > 207.16.137.36.sip: SIP, length: 633 REGISTER sip:207.16.137.36;transport=udp SIP/2.0 Via: SIP/2.0/UDP 207.4.223.35:5080;rport;branch=z9hG4bK9H0SmN8jvavXp Max-Forwards: 70 From: <sip: 6172384723(a)172.16.12.100;transport=udp>;tag=3jXBNmvUZ3XDe To: <sip: 6172384723(a)172.16.12.100;transport=udp> Here is my gateway config <gateway name="voip1"> <param name="username" value="6172384723" /> <param name="password" value="testme" /> <param name="realm" value="172.16.12.100" /> <param name="proxy" value="207.16.137.36"/> <param name="register-proxy" value="207.16.137.36"/> <param name="register" value="true" /> <param name="stun-enabled" value="false"/> <param name="from-domain" value="172.16.12.100"/> <param name="aggressive-nat-detection" value="false"/> </gateway> Here is the bridge from the dialplan <action application="bridge" data="sofia/gateway/voip1/1$1"/> I'm currently running FreeSWITCH Version 1.0.trunk (13226M). I hunted around trying to find another option in for the gateway config that would set the to domain to the realm and still send the packet to proxy but could not find one. Any help/hints are welcome. Thanks, -Dale

2 2

iLBC codec 97 or 102
by seven 08 May '09

08 May '09

Hi, We have FreesSWITCH runing on version 11066, which INVITE with iLBC@30i number 102, however the recent version INVITE with iLBC@30i number 97. Unfortunately I have a broken sip client only talks with RTP payload 102. So here is the question: 1) What's the difference between 97 and 102? The default value changed since version 11066? 2) Is it configurable? Is it part of sofia-lib or FS? Thank you.

3 6

voicemail webapi returns 403 Forbidden
by seven 08 May '09

08 May '09

I loaded mod_xml_rpc on trunk version 13174, to get voicemail, I go to http://192.168.1.27:8080/api/voicemail/web in the challenge box, no matter I input 1009, 1009(a)192.168.1.27, 1009@localhost, 1009@default, I got Error 403. However I use freeswitch/works can get the voicemail interface. but shows 0 messages. However there should be some message. What's wrong of me? it's almost the default config. Thanks. sh-3.2# find storage/ storage/ storage//voicemail storage//voicemail/default storage//voicemail/default/192.168.1.27 storage//voicemail/default/192.168.1.27/1001 storage//voicemail/default/192.168.1.27/1001/ msg_fb314198-7ff8-48a5-957e-cd75885b34ac.wav storage//voicemail/default/192.168.1.27/1008 storage//voicemail/default/192.168.1.27/1009 storage//voicemail/default/192.168.1.27/1009/msg_d5c93a31-0765-41c9- b4e4-ba3493bb5dc5.wav sqlite> select * from voicemail_msgs; 1240973763|1241405588|1009|192.168.1.27|9e5decb2-bb4a-4287-b85f- ddb935878944|Extension 1001|1001|inbox|/usr/local/fs-090428/storage/ voicemail/default/192.168.1.27/1009/msg_d5c93a31-0765-41c9-b4e4- ba3493bb5dc5.wav|15|save|A_URGENT

2 2

Re-2: Ruby and ESL help
by Guido Kuth 08 May '09

08 May '09

Hello Diego, I don't know ruby but I was playing around with outbound socket as well. You have to start your TCPServer and then listen for connections on port 8084 (if you want it like it is standard). If the TCPServer gets a connect request from FS you have to Accept the connection. In .NET this is TCPServer.Accept(). This Returns a TCPClient Object which represents a dedicated connection for this specific call. A new call creates a new TCPClient Object. After that you first have to send a Connect Message ("Connect\n\n") to FS. FS will answer immediately with all data belongig to the call. If this all ist done you can send an Answer command and/or whatever you want. Hope this helps...Guido Btw.: If you find out how one can handle real blocked execution of commands I would like to know how. I tried to playback a long file and my problem was that FS answers immediately after FS accepts the command to play this file, but there is nothing that will ever give you a notice about the playback has ended, what is an unsolved problem for me. -------- Original Message -------- Subject: Re: [Freeswitch-users] Ruby and ESL help (03-Mai-2009 2:23) From: Diego Viola <diego.viola(a)gmail.com> To: gk(a)exram.de > I'm trying to do Event socket outbound btw. > > On Sat, May 2, 2009 at 7:57 PM, Diego Viola <diego.viola(a)gmail.com> wrote: > > Hello everyone, > > > > I was trying to test ESL with Ruby, and I made this: > > > > " > > require 'socket' > > require 'ESL' > > > > TCPServer.new('127.0.0.1', '8084') > > con = ESL::ESLconnection.new('127.0.0.1', '8084', '') > > con.execute('answer') > > con.execute('playback', > > '/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav') > > " > > > > I can connect from freeswitch with sync and async mode, but it doesn't > > do anything more than that, it doesn't execute my answer or playback, > > anyone knows what's wrong with it? I use the freeswitcher lib and it > > works great, but I also want to try ESL. > > > > Thanks, > > > > Diego > > > > _______________________________________________ > 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

5 18

Double Re-Register problem
by Peter P GMX 07 May '09

07 May '09

Hello, I habe the following problem when re-registering to an external SIP provider during a call which results in immediate call-hangups. - FS re-registers with nonce - 2ms later FS re-registers without nonce - external SIP provider asks for credentials - FS re-registers with nonce - External provider hangs up call I think the external equipment (Huawei) gets his messages into disorder and then hangs up. My question is: How can I force FS to only register once (without nonce)? As said, FS tries to register twice within 2 msecs without receiving an answer in between. FS is on a public IP, so there are no NAT problems expected (I can see that until the re-register takes place, media is passed in both directions). Best regards Peter See log: U 2009/05/07 15:04:37.441636 217.xxx.xxx.190:5080 -> 213.xxx.xxx.2:5060 REGISTER sip:sip.provider.de;transport=udp SIP/2.0. Via: SIP/2.0/UDP 217.xxx.xxx.190:5080;rport;branch=z9hG4bK43088Ha5QpZBN. Max-Forwards: 70. From: <sip:0123456779@sip.provider.de;transport=udp>;tag=jSFF9XmFZ50pp. To: <sip:0123456779@sip.provider.de;transport=udp>. Call-ID: a0364af0-3b05-11de-bc92-f9fefd954b7b. CSeq: 114732262 REGISTER. Contact: <sip:gw+XXX_01234@217.xxx.xxx.190:5080;transport=udp>. Expires: 0. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13231M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Authorization: Digest username="0123456779", realm="provider.de", nonce="4a02dc5ba90f927c74161f89e7550138b93f12cc", cnonce="y8SXDLWpEiyNPQAekEzDTg", algorithm=MD5, uri="sip:sip.provider.de;transport=udp", response="1d44f64eec5a5b38b44e398dea201a08", qop=auth, nc=00000002. Content-Length: 0. . # U 2009/05/07 15:04:37.443395 217.xxx.xxx.190:5080 -> 213.xxx.xxx.2:5060 REGISTER sip:sip.provider.de;transport=udp SIP/2.0. Via: SIP/2.0/UDP 217.xxx.xxx.190:5080;rport;branch=z9hG4bK5ct1aDU8mZNyg. Max-Forwards: 70. From: <sip:0123456779@sip.provider.de;transport=udp>;tag=t1SNQpUB8cKcK. To: <sip:0123456779@sip.provider.de;transport=udp>. Call-ID: a0364af0-3b05-11de-bc92-f9fefd954b7b. CSeq: 114732402 REGISTER. Contact: <sip:gw+XXX_01234@217.xxx.xxx.190:5080;transport=udp>. Expires: 60. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13231M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Content-Length: 0. . # U 2009/05/07 15:04:37.466691 213.xxx.xxx.2:5060 -> 217.xxx.xxx.190:5080 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 217.xxx.xxx.190:5080;branch=z9hG4bK5ct1aDU8mZNyg;rport=5080. Call-ID: a0364af0-3b05-11de-bc92-f9fefd954b7b. From: <sip:0123456779@sip.provider.de;transport=udp>;tag=t1SNQpUB8cKcK. To: <sip:0123456779@sip.provider.de;transport=udp>;tag=702dbe10. CSeq: 114732402 REGISTER. Server: SIP Router. WWW-Authenticate: Digest realm="provider.de",nonce="4a02dd789a25b67f29ba21f65429d13c4bbc2ded",qop="auth". Content-Length: 0. . # U 2009/05/07 15:04:37.467211 217.xxx.xxx.190:5080 -> 213.xxx.xxx.2:5060 REGISTER sip:sip.provider.de;transport=udp SIP/2.0. Via: SIP/2.0/UDP 217.xxx.xxx.190:5080;rport;branch=z9hG4bK6NKtc8Bcj8BHc. Max-Forwards: 70. From: <sip:0123456779@sip.provider.de;transport=udp>;tag=t1SNQpUB8cKcK. To: <sip:0123456779@sip.provider.de;transport=udp>. Call-ID: a0364af0-3b05-11de-bc92-f9fefd954b7b. CSeq: 114732403 REGISTER. Contact: <sip:gw+XXX_01234@217.xxx.xxx.190:5080;transport=udp>. Expires: 60. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13231M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Authorization: Digest username="0123456779", realm="provider.de", nonce="4a02dd789a25b67f29ba21f65429d13c4bbc2ded", cnonce="dbAgB7WqEiyNPQAekEzDTg", algorithm=MD5, uri="sip:sip.provider.de;transport=udp", response="6a55b27caec6b06bd9da707e7b24d82b", qop=auth, nc=00000001. Content-Length: 0. . # U 2009/05/07 15:04:37.470935 213.xxx.xxx.2:5060 -> 217.xxx.xxx.190:5080 BYE sip:gw+XXX_01234@217.xxx.xxx.190:5080;transport=udp SIP/2.0. Via: SIP/2.0/UDP 213.xxx.xxx.2:5060;branch=z9hG4bK400739ec3ad9e5bbd7f5edccf. Call-ID: d0e38021-b5a9-122c-3d8d-001e904cc34e. From: <sip:0049987654321@sip.provider.de>;tag=31de8a21. To: "unknown"<sip:0123456779@sip.provider.de;transport=udp>;tag=K287aS5jveQ9H. CSeq: 1 BYE. Max-Forwards: 70. Content-Length: 0.

2 3

Originate - using own uuid
by Simon Tang 07 May '09

07 May '09

Hi there, I know there's a "feature" that allows us to create a uuid and then use that uuid to do an originate (api create_uuid). Is there any badness if I don't do the "create_uuid" and just do the originate using my own uuid. I experimented and made the following originate call: api originate {origination_caller_id_number=16041234567,originate_timeout=60,originati on_uuid=abcdefg}sofia/gateway/icall/16041234567 &park ...and this works, but just want to make sure if there's any badness of doing so...that is, if I can guarantee that the origination_uuid I will be passing in *WILL* be unique.

4 6

Amazon EC2 no audio
by Dave Grootwassink 07 May '09

07 May '09

Hello all, Help a n00b out. I have been trying to get an instance of FreeSwitch running up in the Amazon EC2 cloud. I have successfully gotten the package built following the wiki and archives of this list. I can get x-lite to register with the switch and it will set up calls out on my asterlink account. The problem is that there is no audio transfer (so I am assuming RTP problem). The setup: Firewall open ports tcp 0-65535 udp 0-65535 --- I tried so many combinations unsuccessfully, I finally just blasted open everything. In conf/freeswitch.xml (174.129.201.96 is assigned elastic IP address) <X-PRE-PROCESS cmd="include" data="vars.xml"/> #set "external_rtp_ip=174.129.201.96" <section name="configuration" description="Various Configuration"> <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml"/> </section> In conf/autoload_configs/sofia.conf.xml <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_rtp_ip}"/> Internal network IP assignment: Name: domU-12-31-39-00-84-B6.compute-1.internal Address: 10.254.139.68 When I setup a call through asterlink, I see this in the system log: Ring SDP: v=0 o=FreeSWITCH 1241648830 1241648831 IN IP4 10.254.139.68 s=FreeSWITCH c=IN IP4 10.254.139.68 t=0 0 m=audio 17654 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 a=sendrecv I do not know SDP parameters off the top of my head, but I am assuming that it is telling Asterlink to route the RTP to the internal network IP address and not the external one.

2 1

FS + encryption
by Paul 07 May '09

07 May '09

Yes, I've seen this http://wiki.freeswitch.org/wiki/SIP_TLS. I was just curious if the only way to have true end to end secure communications with FS would have to be a SIP trunk from one FS system to another encrypted SIP system on the other with no POTS/PRI/BRI circuits used in transit. I'm assuming if there's any POTS/BRI/PRI/DSS circuits used in transit, anyone with a lineman's handset could still eavesdrop on any conversations. Is this not the case? Paul

4 5