I am new to free switch, and I am pretty amazed by the features and potentials of it. After reading a lot from wiki site and lists, I managed to make the freeswitch rc2 up and running on my centos 5, and had the sip extension 1000 on Bria 2.1and sip extension 1001 on xlite 3.0 registered from my XP PC to the freeswitch on centos, also have four extensions 1004-1007 on Grandstream 4004 and two extensions 1010 and 1011 on PAP2-NA all registered on freeswicth, I was able to hear music on hold by dialing 9999 and joined conference 300x, I could call each other and saw presence of the extensions on Bria 2.1 , and the extensions could answer the ringing. But I coudn't hear the remote calling party after answering each of the call, I only heard the voice mail prompt, when I dialed any number 1 or 2 or 3 or any digit, seemed no DTMF detection and no audio being sent. Could you guys shed some light on this and put me to the right direction? Your guidance is much appreciated. Thanks Chris
This smells of a nat related issue. Can you explain the network layout and where nat is involved. Mike On Apr 10, 2008, at 1:32 PM, Chris Chen wrote:
I am new to free switch, and I am pretty amazed by the features and potentials of it.
After reading a lot from wiki site and lists, I managed to make the freeswitch rc2 up and running on my centos 5, and had the sip extension 1000 on Bria 2.1and sip extension 1001 on xlite 3.0 registered from my XP PC to the freeswitch on centos, also have four extensions 1004-1007 on Grandstream 4004 and two extensions 1010 and 1011 on PAP2-NA all registered on freeswicth, I was able to hear music on hold by dialing 9999 and joined conference 300x, I could call each other and saw presence of the extensions on Bria 2.1 , and the extensions could answer the ringing.
But I coudn't hear the remote calling party after answering each of the call, I only heard the voice mail prompt, when I dialed any number 1 or 2 or 3 or any digit, seemed no DTMF detection and no audio being sent.
Could you guys shed some light on this and put me to the right direction?
Your guidance is much appreciated.
Thanks
Chris _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks Mike, they are all in the same network 10.0.1.x Freeswitch 10.0.1.252 XP PC 10.0.1.253 GXW4004 10.0.1.160 PAP2-NA 10.0.1.251 I am using the default profile, with the SIP-IP and RTP-IP changed to 10.0.1.252 Do you have any specific catch in any of these settings, sofia.conf.xml or in sip profile default.xml? Thanks Chris On Thu, Apr 10, 2008 at 2:18 PM, Michael Jerris <mike@jerris.com> wrote:
This smells of a nat related issue. Can you explain the network layout and where nat is involved.
Mike
On Apr 10, 2008, at 1:32 PM, Chris Chen wrote:
I am new to free switch, and I am pretty amazed by the features and potentials of it.
After reading a lot from wiki site and lists, I managed to make the freeswitch rc2 up and running on my centos 5, and had the sip extension 1000 on Bria 2.1and sip extension 1001 on xlite 3.0 registered from my XP PC to the freeswitch on centos, also have four extensions 1004-1007 on Grandstream 4004 and two extensions 1010 and 1011 on PAP2-NA all registered on freeswicth, I was able to hear music on hold by dialing 9999 and joined conference 300x, I could call each other and saw presence of the extensions on Bria 2.1 , and the extensions could answer the ringing.
But I coudn't hear the remote calling party after answering each of the call, I only heard the voice mail prompt, when I dialed any number 1 or 2 or 3 or any digit, seemed no DTMF detection and no audio being sent.
Could you guys shed some light on this and put me to the right direction?
Your guidance is much appreciated.
Thanks
Chris _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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Without the change, the domain name and rtp-ip sip-ip would be 127.0.0.1which is the loopback IP address of the freeswitch server. After those changes, at least I had all sip end-points registered and I heard dialtones when off the hook. Thanks On Thu, Apr 10, 2008 at 2:31 PM, Brian West <brian@freeswitch.org> wrote:
Can you elaborate on why you did this change? /b
On Apr 10, 2008, at 1:25 PM, Chris Chen wrote:
I am using the default profile, with the SIP-IP and RTP-IP changed to 10.0.1.252
Brian West sip:brian@freeswitch.org
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OK just need to know if I had a bug in the default config. /b On Apr 10, 2008, at 1:37 PM, Chris Chen wrote:
Without the change, the domain name and rtp-ip sip-ip would be 127.0.0.1 which is the loopback IP address of the freeswitch server. After those changes, at least I had all sip end-points registered and I heard dialtones when off the hook. Thanks
Brian West sip:brian@freeswitch.org
Just tossing this out there. Are iptables running? as root run: `iptables -L` Make sure that ports 5060-5080 (tcp and udp) are not being blocked. *Chris Danielson* chris@maxpowersoft.com <mailto:chris@maxpowersoft.com> Chris Chen wrote:
Without the change, the domain name and rtp-ip sip-ip would be 127.0.0.1 <http://127.0.0.1> which is the loopback IP address of the freeswitch server. After those changes, at least I had all sip end-points registered and I heard dialtones when off the hook. Thanks
On Thu, Apr 10, 2008 at 2:31 PM, Brian West <brian@freeswitch.org <mailto:brian@freeswitch.org>> wrote:
Can you elaborate on why you did this change?
/b
On Apr 10, 2008, at 1:25 PM, Chris Chen wrote:
I am using the default profile, with the SIP-IP and RTP-IP changed to 10.0.1.252 <http://10.0.1.252>
Brian West sip:brian@freeswitch.org
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------------------------------------------------------------------------
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Firewall is disabled on the centos 5 as this is purely LAN environment to start with. I believe there is no signaling issue here as every SIP endpoint is able to be registered and with presence information. The issue is about the media can be heard but not able to be sent out. on the sip endpoints. Thanks Chris On Thu, Apr 10, 2008 at 2:49 PM, Chris Danielson <chris@maxpowersoft.com> wrote:
Just tossing this out there. Are iptables running? as root run: `iptables -L`
Make sure that ports 5060-5080 (tcp and udp) are not being blocked.
*Chris Danielson* chris@maxpowersoft.com
Chris Chen wrote:
Without the change, the domain name and rtp-ip sip-ip would be 127.0.0.1which is the loopback IP address of the freeswitch server. After those changes, at least I had all sip end-points registered and I heard dialtones when off the hook. Thanks
On Thu, Apr 10, 2008 at 2:31 PM, Brian West <brian@freeswitch.org> wrote:
Can you elaborate on why you did this change? /b
On Apr 10, 2008, at 1:25 PM, Chris Chen wrote:
I am using the default profile, with the SIP-IP and RTP-IP changed to 10.0.1.252
Brian West sip:brian@freeswitch.org
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------------------------------
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Hi Chris, Your problem seems a lot like what's happening to me, although I'm only using xlite softphones on extensions 1000 and 1001. The phones are able to register and make calls (well two calls at least, before FS stops responding), but the media is not getting through. I'm also using CentOS 5, but the server and softphone computers are on public IPs connected directly to the Internet, and there's no firewall running. I made a test of registering with extension 1000 and calling the number 9998 2 times, the first I can hear 3 secs of the tetris sound, and hanging up on the softphone ends the call, but on the second call I get no audio at all, and when I hang up on the softphone, the call is not ended on FS, so it stays open for a while until it times out and FS starts deperately trying to send BYE messages to the softphone. Here are my logs and sip trace (user/pass: freeswitch/mailing): - http://www.medularis.com/fs/freeswitch.log - http://www.medularis.com/fs/consolelog.txt - http://www.medularis.com/fs/siptrace.log I guess I'm not being very useful to you in solving the problem, but I thought the issues could be related, so maybe we could be able to get the same help. On 4/10/08, Chris Chen <chris.chen2004@gmail.com> wrote:
Firewall is disabled on the centos 5 as this is purely LAN environment to start with. I believe there is no signaling issue here as every SIP endpoint is able to be registered and with presence information. The issue is about the media can be heard but not able to be sent out. on the sip endpoints.
Thanks
Chris
On Thu, Apr 10, 2008 at 2:49 PM, Chris Danielson <chris@maxpowersoft.com> wrote:
Just tossing this out there. Are iptables running? as root run: `iptables -L`
Make sure that ports 5060-5080 (tcp and udp) are not being blocked.
Chris Danielson chris@maxpowersoft.com
Chris Chen wrote:
Without the change, the domain name and rtp-ip sip-ip would be 127.0.0.1
which is the loopback IP address of the freeswitch server. After those changes, at least I had all sip end-points registered and I heard dialtones when off the hook.
Thanks
On Thu, Apr 10, 2008 at 2:31 PM, Brian West <brian@freeswitch.org> wrote:
Can you elaborate on why you did this change?
/b
On Apr 10, 2008, at 1:25 PM, Chris Chen wrote:
I am using the default profile, with the SIP-IP and RTP-IP changed to
10.0.1.252
Brian West sip:brian@freeswitch.org
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Regards, -- Nicolás Brenner
A follow up to this issue: I resolved this issue by setting the ext-sip-ip and ext-rtp-ip to 10.0.1.252. in the vars.xml as this freeswitch is going nowhere to the internet. So basically this one way audio issue is the NAT issue, the typical SIP issue. now the sip default profile looks like this: freeswitch@callcenter> sofia status profile default API CALL [sofia(status profile default)] output: ================================================================================================= Name default Domain Name 10.0.1.252 DBName sofia_reg_default Dialplan XML RTP-IP 10.0.1.252 Ext-RTP-IP 10.0.1.252 SIP-IP 10.0.1.252 Ext-SIP-IP 10.0.1.252 URL sip:mod_sofia@10.0.1.252:5060 BIND-URL sip:mod_sofia@10.0.1.252:5060;maddr=10.0.1.252 HOLD-MUSIC local_stream://moh CODECS G722,PCMU,PCMA,GSM,H263-1998 TEL-EVENT 101 CNG 13 SESSION-TO 0 MAX-DIALOG 0 Registrations: ================================================================================================= Call-ID Zjg0YTVjMmU0NzRhN2NhZjA1MmNiOTEyZWExN2M4YTY. User 1000@10.0.1.252 Contact "1000" <sip:1000@10.0.1.253:36784 ;rinstance=6f9ee3cff586843a> Agent Bria release 2.2 stamp 45414 Status Registered(unknown) EXP(2008-04-11 00:15:24) Call-ID 1858798975-5064-1@10.0.1.160 User 1006@10.0.1.252 Contact "user" <sip:1006@10.0.1.160:5064> Agent Grandstream GXW-4004 V1.1A 1.0.0.67 Status Registered(unknown) EXP(2008-04-11 00:20:37) Call-ID 1412946610-5066-1@10.0.1.160 User 1007@10.0.1.252 Contact "user" <sip:1007@10.0.1.160:5066> Agent Grandstream GXW-4004 V1.1A 1.0.0.67 Status Registered(unknown) EXP(2008-04-11 00:20:37) Call-ID 51e43082-5491ec6d@10.0.1.251 User 1010@10.0.1.252 Contact 1010 <sip:1010@10.0.1.251:5060> Agent Linksys/PAP2-3.1.12(LS) Status Registered(unknown) EXP(2008-04-11 00:21:11) Call-ID f37727a1-518ac7da@10.0.1.251 User 1011@10.0.1.252 Contact 1011 <sip:1011@10.0.1.251:5061> Agent Linksys/PAP2-3.1.12(LS) Status Registered(unknown) EXP(2008-04-11 00:21:11) Call-ID 445786933-5060-1@10.0.1.160 User 1004@10.0.1.252 Contact "user" <sip:1004@10.0.1.160:5060> Agent Grandstream GXW-4004 V1.1A 1.0.0.67 Status Registered(unknown) EXP(2008-04-11 00:21:21) Call-ID 1697459759-5062-1@10.0.1.160 User 1005@10.0.1.252 Contact "user" <sip:1005@10.0.1.160:5062> Agent Grandstream GXW-4004 V1.1A 1.0.0.67 Status Registered(unknown) EXP(2008-04-11 00:21:21) ================================================================================================= I really appreciate your valuable advice to me for the happy freeswitch journey. Lots of good stuffs to be explored further. Thanks, Chris On Thu, Apr 10, 2008 at 3:16 PM, Nicolas Brenner <nicolas@medularis.com> wrote:
Hi Chris,
Your problem seems a lot like what's happening to me, although I'm only using xlite softphones on extensions 1000 and 1001. The phones are able to register and make calls (well two calls at least, before FS stops responding), but the media is not getting through. I'm also using CentOS 5, but the server and softphone computers are on public IPs connected directly to the Internet, and there's no firewall running.
I made a test of registering with extension 1000 and calling the number 9998 2 times, the first I can hear 3 secs of the tetris sound, and hanging up on the softphone ends the call, but on the second call I get no audio at all, and when I hang up on the softphone, the call is not ended on FS, so it stays open for a while until it times out and FS starts deperately trying to send BYE messages to the softphone.
Here are my logs and sip trace (user/pass: freeswitch/mailing): - http://www.medularis.com/fs/freeswitch.log - http://www.medularis.com/fs/consolelog.txt - http://www.medularis.com/fs/siptrace.log
I guess I'm not being very useful to you in solving the problem, but I thought the issues could be related, so maybe we could be able to get the same help.
On 4/10/08, Chris Chen <chris.chen2004@gmail.com> wrote:
Firewall is disabled on the centos 5 as this is purely LAN environment to start with. I believe there is no signaling issue here as every SIP endpoint is able to be registered and with presence information. The issue is about the media can be heard but not able to be sent out. on the sip endpoints.
Thanks
Chris
On Thu, Apr 10, 2008 at 2:49 PM, Chris Danielson <chris@maxpowersoft.com
wrote:
Just tossing this out there. Are iptables running? as root run: `iptables -L`
Make sure that ports 5060-5080 (tcp and udp) are not being blocked.
Chris Danielson chris@maxpowersoft.com
Chris Chen wrote:
Without the change, the domain name and rtp-ip sip-ip would be
127.0.0.1 which is the loopback IP address of the freeswitch server. After those changes, at least I had all sip end-points registered and I heard dialtones when off the hook.
Thanks
On Thu, Apr 10, 2008 at 2:31 PM, Brian West <brian@freeswitch.org> wrote:
Can you elaborate on why you did this change?
/b
On Apr 10, 2008, at 1:25 PM, Chris Chen wrote:
I am using the default profile, with the SIP-IP and RTP-IP changed
to 10.0.1.252
Brian West sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org>
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Regards,
-- Nicolás Brenner
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participants (5)
-
Brian West
-
Chris Chen
-
Chris Danielson
-
Michael Jerris
-
Nicolas Brenner