FS 1.0.6 on EC2 - no audio on outgoing calls
Hi there, I am experimenting with FS on EC2, I like results, but stuck on weird audio issue - I followed FreeSwitch EC2 wiki article and modified internal profile and vars.xml accordingly, but unfortunately still cannot get it working. Incoming and outgoing calls made using a SIP phone to FS extensions work just fine. As well as calls to FS from PSTN. But calls to PSTN via gateways result in no audio at all, no ring, nothing, SIP signaling goes through OK. Sofia status profile shows correct values for Ext-RTP-IP for both profiles - my static public IP, RTP-IP shows local IP. Any thoughts on that? Anybody can share working profile configuration may be? Please help, I really need to get this going. Thanks.
firewall or iptables On Tue, Jul 6, 2010 at 8:02 PM, paul gore <paul.gore.j@gmail.com> wrote:
Hi there, I am experimenting with FS on EC2, I like results, but stuck on weird audio issue - I followed FreeSwitch EC2 wiki article and modified internal profile and vars.xml accordingly, but unfortunately still cannot get it working. Incoming and outgoing calls made using a SIP phone to FS extensions work just fine. As well as calls to FS from PSTN. But calls to PSTN via gateways result in no audio at all, no ring, nothing, SIP signaling goes through OK. Sofia status profile shows correct values for Ext-RTP-IP for both profiles - my static public IP, RTP-IP shows local IP. Any thoughts on that? Anybody can share working profile configuration may be? Please help, I really need to get this going.
Thanks.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one? On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote:
Hi there, I am experimenting with FS on EC2, I like results, but stuck on weird audio issue - I followed FreeSwitch EC2 wiki article and modified internal profile and vars.xml accordingly, but unfortunately still cannot get it working. Incoming and outgoing calls made using a SIP phone to FS extensions work just fine. As well as calls to FS from PSTN. But calls to PSTN via gateways result in no audio at all, no ring, nothing, SIP signaling goes through OK. Sofia status profile shows correct values for Ext-RTP-IP for both profiles - my static public IP, RTP-IP shows local IP. Any thoughts on that? Anybody can share working profile configuration may be? Please help, I really need to get this going.
Thanks.
not really, not with so little information. On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com> wrote:
Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one?
On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote:
Hi there, I am experimenting with FS on EC2, I like results, but stuck on weird audio issue - I followed FreeSwitch EC2 wiki article and modified internal profile and vars.xml accordingly, but unfortunately still cannot get it working. Incoming and outgoing calls made using a SIP phone to FS extensions work just fine. As well as calls to FS from PSTN. But calls to PSTN via gateways result in no audio at all, no ring, nothing, SIP signaling goes through OK. Sofia status profile shows correct values for Ext-RTP-IP for both profiles - my static public IP, RTP-IP shows local IP. Any thoughts on that? Anybody can share working profile configuration may be? Please help, I really need to get this going.
Thanks.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
I had same problem from this provider without to explain why. One day it works, another day it doesn't, their support is crap... ----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 07, 2010 2:37 PM Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls not really, not with so little information. On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com> wrote: Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one? On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird audio > issue - I followed FreeSwitch EC2 wiki article and modified internal > profile > and vars.xml accordingly, but unfortunately still cannot get it working. > Incoming and outgoing calls made using a SIP phone to FS extensions work > just fine. As well as calls to FS from PSTN. But calls to PSTN via gateways > result in no audio at all, no ring, nothing, SIP signaling goes through OK. > Sofia status profile shows correct values for Ext-RTP-IP for both profiles > - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration may > be? > Please help, I really need to get this going. > > Thanks. > _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 ------------------------------------------------------------------------------ _______________________________________________ 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
What "doesn't work" means? It could be (and most likely is not) FS-related problem On Wednesday 07 July 2010, Madovsky wrote:
I had same problem from this provider without to explain why. One day it works, another day it doesn't, their support is crap...
----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 07, 2010 2:37 PM Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
not really, not with so little information.
On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com> wrote:
Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one?
On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird > audio issue - I followed FreeSwitch EC2 wiki article and modified > internal profile > and vars.xml accordingly, but unfortunately still cannot get it > working. Incoming and outgoing calls made using a SIP phone to FS > extensions work just fine. As well as calls to FS from PSTN. But > calls to PSTN via gateways result in no audio at all, no ring, > nothing, SIP signaling goes through OK. Sofia status profile shows > correct values for Ext-RTP-IP for both profiles - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration > may be? > Please help, I really need to get this going. > > Thanks.
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You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo). On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
What "doesn't work" means? It could be (and most likely is not) FS-related problem
On Wednesday 07 July 2010, Madovsky wrote:
I had same problem from this provider without to explain why. One day it works, another day it doesn't, their support is crap...
----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 07, 2010 2:37 PM Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
not really, not with so little information.
On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com> wrote:
Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one?
On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird > audio issue - I followed FreeSwitch EC2 wiki article and modified > internal profile > and vars.xml accordingly, but unfortunately still cannot get it > working. Incoming and outgoing calls made using a SIP phone to FS > extensions work just fine. As well as calls to FS from PSTN. But > calls to PSTN via gateways result in no audio at all, no ring, > nothing, SIP signaling goes through OK. Sofia status profile shows > correct values for Ext-RTP-IP for both profiles - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration > may be? > Please help, I really need to get this going. > > Thanks.
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-- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
This provider does work on another box which is not natted as ec2. Most puzzling here though is why call originaion via api even not going via siptraffic still gets no audio. On 7/7/10, Tony Graziano <tgraziano@myitdepartment.net> wrote:
You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo).
On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
What "doesn't work" means? It could be (and most likely is not) FS-related problem
On Wednesday 07 July 2010, Madovsky wrote:
I had same problem from this provider without to explain why. One day it works, another day it doesn't, their support is crap...
----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 07, 2010 2:37 PM Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
not really, not with so little information.
On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com> wrote:
Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one?
On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird > audio issue - I followed FreeSwitch EC2 wiki article and modified > internal profile > and vars.xml accordingly, but unfortunately still cannot get it > working. Incoming and outgoing calls made using a SIP phone to FS > extensions work just fine. As well as calls to FS from PSTN. But > calls to PSTN via gateways result in no audio at all, no ring, > nothing, SIP signaling goes through OK. Sofia status profile shows > correct values for Ext-RTP-IP for both profiles - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration > may be? > Please help, I really need to get this going. > > Thanks.
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-- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427
Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
So "don't work" still means "no audio" and "sip is ok", clear... RTP packets are lost somewhere on the way: You can check where RTP packets are lost - between FS and siptraffic, or between FS and your SIP Client. 1.Install "ngrep" if you don't have it yet (on Ubuntu: apt-get install ngrep) 2.Run it: "ngrep port 5080" 3.Start a call and after few ngrep's messages, stop ngrep using Ctrl+C 4.Find the first INVITE line from your FS to siptraffic, it will contain your IP and port at the following lines: c=IN IP4 ****!!!YOUR-EC-IP!!!*****..t=0 0..m=audio ****!!!YOUR-RTP-PORT!!!**** RTP/AVP 5.Open new terminal window, and run "ngrep port ****!!!YOUR-RTP-PORT!!!****" 6. Wait 5 seconds 7. Stop ngrep using Ctrl+C 8. Hangup Now on the second terminal you should see a lot of line pairs like: YOUR-EC-IP -> SIPTRAFFIC-IP SIPTRAFFIC-IP -> YOUR-EC-IP If you see only one of the directions (e.g. only YOUR-EC-IP -> SIPTRAFFIC-IP), then some problem is between FS and Siptraffic. If you see both directions then problem is not here and most probably on the way from FS to your SIP Client or somewhere else (inside FS?) If so, try to investigate this part using port 5060 (same way as 5080). This analysis will narrow the problem a bit... Regads, Vitalie 2010/7/7 paul gore <paul.gore.j@gmail.com>
This provider does work on another box which is not natted as ec2. Most puzzling here though is why call originaion via api even not going via siptraffic still gets no audio.
On 7/7/10, Tony Graziano <tgraziano@myitdepartment.net> wrote:
You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo).
On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
What "doesn't work" means? It could be (and most likely is not) FS-related problem
On Wednesday 07 July 2010, Madovsky wrote:
I had same problem from this provider without to explain why. One day it works, another day it doesn't, their support is crap...
----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 07, 2010 2:37 PM Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
not really, not with so little information.
On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com> wrote:
Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one?
On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird > audio issue - I followed FreeSwitch EC2 wiki article and modified > internal profile > and vars.xml accordingly, but unfortunately still cannot get it > working. Incoming and outgoing calls made using a SIP phone to FS > extensions work just fine. As well as calls to FS from PSTN. But > calls to PSTN via gateways result in no audio at all, no ring, > nothing, SIP signaling goes through OK. Sofia status profile shows > correct values for Ext-RTP-IP for both profiles - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration > may be? > Please help, I really need to get this going. > > Thanks.
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-- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427
Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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May be it's just a FAS? Are you able to call that number with a different SIP client using the same call termination provider? On Wednesday 07 July 2010, Vitalii Colosov wrote:
So "don't work" still means "no audio" and "sip is ok", clear...
RTP packets are lost somewhere on the way:
You can check where RTP packets are lost - between FS and siptraffic, or between FS and your SIP Client.
1.Install "ngrep" if you don't have it yet (on Ubuntu: apt-get install ngrep) 2.Run it: "ngrep port 5080" 3.Start a call and after few ngrep's messages, stop ngrep using Ctrl+C 4.Find the first INVITE line from your FS to siptraffic, it will contain your IP and port at the following lines: c=IN IP4 ****!!!YOUR-EC-IP!!!*****..t=0 0..m=audio ****!!!YOUR-RTP-PORT!!!**** RTP/AVP 5.Open new terminal window, and run "ngrep port ****!!!YOUR-RTP-PORT!!!****" 6. Wait 5 seconds 7. Stop ngrep using Ctrl+C 8. Hangup
Now on the second terminal you should see a lot of line pairs like: YOUR-EC-IP -> SIPTRAFFIC-IP SIPTRAFFIC-IP -> YOUR-EC-IP
If you see only one of the directions (e.g. only YOUR-EC-IP -> SIPTRAFFIC-IP), then some problem is between FS and Siptraffic.
If you see both directions then problem is not here and most probably on the way from FS to your SIP Client or somewhere else (inside FS?) If so, try to investigate this part using port 5060 (same way as 5080).
This analysis will narrow the problem a bit...
Regads, Vitalie
2010/7/7 paul gore <paul.gore.j@gmail.com>
This provider does work on another box which is not natted as ec2. Most puzzling here though is why call originaion via api even not going via siptraffic still gets no audio.
On 7/7/10, Tony Graziano <tgraziano@myitdepartment.net> wrote:
You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo).
On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin
<sos@sokhapkin.dyndns.org>wrote:
What "doesn't work" means? It could be (and most likely is not)
FS-related
problem
On Wednesday 07 July 2010, Madovsky wrote:
I had same problem from this provider without to explain why. One day it works, another day it doesn't, their support is crap...
----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 07, 2010 2:37 PM Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on
outgoing
calls
not really, not with so little information.
On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com>
wrote:
Firewall is configured according to the wiki, I also tried to open
all
udp ports, issue persists. Actually the problem became more complex - outgoing calls don't
work
with one particular termination provider, siptraffic.com , any
ideas
why? Outgoing calls also don't work when originating a call via js
script
or via FS api. Any clues on that one?
On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck > on
weird
> audio issue - I followed FreeSwitch EC2 wiki article and
modified
> internal profile > and vars.xml accordingly, but unfortunately still cannot get > it working. Incoming and outgoing calls made using a SIP phone > to
FS
> extensions work just fine. As well as calls to FS from PSTN. > But calls to PSTN via gateways result in no audio at all, no > ring, nothing, SIP signaling goes through OK. Sofia status > profile
shows
> correct values for Ext-RTP-IP for both profiles - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile
configuration
> may be? > Please help, I really need to get this going. > > Thanks.
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-- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427
Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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Seems like siptraffic uses 6 ip addresses for media, can that be the problem? Is there any setting in a gateway config xml which helps with that? I will do ngrep thing and update. On 7/7/10, paul gore <paul.gore.j@gmail.com> wrote:
This provider does work on another box which is not natted as ec2. Most puzzling here though is why call originaion via api even not going via siptraffic still gets no audio.
On 7/7/10, Tony Graziano <tgraziano@myitdepartment.net> wrote:
You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo).
On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
What "doesn't work" means? It could be (and most likely is not) FS-related problem
On Wednesday 07 July 2010, Madovsky wrote:
I had same problem from this provider without to explain why. One day it works, another day it doesn't, their support is crap...
----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 07, 2010 2:37 PM Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
not really, not with so little information.
On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com> wrote:
Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one?
On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird > audio issue - I followed FreeSwitch EC2 wiki article and modified > internal profile > and vars.xml accordingly, but unfortunately still cannot get it > working. Incoming and outgoing calls made using a SIP phone to FS > extensions work just fine. As well as calls to FS from PSTN. But > calls to PSTN via gateways result in no audio at all, no ring, > nothing, SIP signaling goes through OK. Sofia status profile shows > correct values for Ext-RTP-IP for both profiles - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration > may be? > Please help, I really need to get this going. > > Thanks.
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-- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427
Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
There should not be any problem if they use multiple IP addresses for media stream, as long as siptraffic's SDP contains correct IP that will participate in RTPs exchange... Do note that media IP can be different from sip IP, and it is OK, if implemented properly... 2010/7/8 paul gore <paul.gore.j@gmail.com>
Seems like siptraffic uses 6 ip addresses for media, can that be the problem? Is there any setting in a gateway config xml which helps with that? I will do ngrep thing and update.
On 7/7/10, paul gore <paul.gore.j@gmail.com> wrote:
This provider does work on another box which is not natted as ec2. Most puzzling here though is why call originaion via api even not going via siptraffic still gets no audio.
On 7/7/10, Tony Graziano <tgraziano@myitdepartment.net> wrote:
You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo).
On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
What "doesn't work" means? It could be (and most likely is not) FS-related problem
On Wednesday 07 July 2010, Madovsky wrote:
I had same problem from this provider without to explain why. One day it works, another day it doesn't, their support is crap...
----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 07, 2010 2:37 PM Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
not really, not with so little information.
On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com> wrote:
Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one?
On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird > audio issue - I followed FreeSwitch EC2 wiki article and modified > internal profile > and vars.xml accordingly, but unfortunately still cannot get it > working. Incoming and outgoing calls made using a SIP phone to FS > extensions work just fine. As well as calls to FS from PSTN. But > calls to PSTN via gateways result in no audio at all, no ring, > nothing, SIP signaling goes through OK. Sofia status profile shows > correct values for Ext-RTP-IP for both profiles - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration > may be? > Please help, I really need to get this going. > > Thanks.
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-- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427
Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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I got ngrep trace for port 5060 while making a call to a US number via siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I heard no audio not even ringing. Is there anything in this trace which can help identify the problem? 10.194.206.102:5060 - is my local EC2 IP 184.72.206.204:5060 - is my public EC2 IP 77.72.169.128:5060 - siptraffic.com proxy IP Thanks! 67.33.160.119:18294 -> 10.194.206.102:5060 INVITE sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f 524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: < sip:4000002@67.33.160.119:18027>..To: "45517 709248570"<sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>..From: "4000002"<sip:4000002@myserver.com <sip%3A4000002@myserver.com>
;tag=5f1ec15f..Call- ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest user name="4000002",realm="myserver.com ",nonce="cf9019cc-f44a-4568-97d1-e9883fb1821f",uri="sip:45517705678570@v ersafon.com ",response="57da72527524e0e065c9a3221bfadd38",cnonce="140f655ff3427f6ba3767ab7040231f3",nc=0000 0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41150..Content-Length: 417....v=0..o=- 8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107 119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8 46298..a=fmtp:101 0-15..a=rtpmap:107 BV32/16000..a=rtpmap:119 BV32-FEC/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap :105 SPEEX-FEC/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r port=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To: "455177092 48570" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I NVITE..User-Agent: myserver..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 INVITE sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204> ;tag=18853e82KDe7j. .To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 INVITE..Contact: <sip:gw+voicetrading.com@184.72.206.204:5080 ;transport=udp;gw=voicetrading.com>..User-A gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.. Supported: timer, precondition, path, replaces..Allow-Events: talk, refer..Content-Type: application/sdp.. Content-Disposition: session..Content-Length: 295..X-FS-Support: update_display..Remote-Party-ID: "4000002 " <sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1278518039 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/80 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20.. # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..From : "4000002" <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.co <sip%3A0017705678570@sip.siptraffic.co> m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 198....v=0..o=C ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4 77.72.168.40..t=0 0..m=audio 57672 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20.. # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1- -d87543-;rport=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To: "45517705678570" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjk zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact: <sip:45517705678570@184.72.206.204:5060;transport=udp>..User-A gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message- summary, refer..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 251..Remote-P arty-ID: "45517705678570" <sip:45517705678570@10.194.206.102<sip%3A45517705678570@10.194.206.102> ;party=calling;privacy=off;screen=no....v=0.. o=FreeSWITCH 1278530815 1278530816 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m= audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=sil enceSupp:off - - - -..a=ptime:20.. #
U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..From : "4000002" <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.co <sip%3A0017705678570@sip.siptraffic.co> m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 204....v=0..o=C ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN IP4 208.167.230.118..t=0 0..m=audio 57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20.. # U 67.33.160.119:18294 -> 10.194.206.102:5060 .... # U 67.33.160.119:18294 -> 10.194.206.102:5060 CANCEL sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f 524431af92cef56-1--d87543-;rport..To: "45517705678570"< sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>>..From: "4000002"<s ip:4000002@myserver.com <ip%3A4000002@myserver.com>>;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC EL..Proxy-Authorization: Digest username="4000002",realm="myserver.com ",nonce="cf9019cc-f44a-4568-97d1-e98 83fb1821f",uri="sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com> ",response="46c7e289f7490c807565c561699b03d6",cnonce="a226c 55446c605ee229f045602b29135",nc=00000002,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41 150..Content-Length: 0.... # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport =18294;received=67.33.160.119..From: "4000002" <sip:4000002@myserver.com<sip%3A4000002@myserver.com>>;tag=5f1ec15f..To: "4551770567857 0" <sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY mY...CSeq: 2 CANCEL..Content-Length: 0.... # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56- 1--d87543-;rport=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To : "45517705678570" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmN jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSA GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, repla ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presen ce.winfo, message-summary, refer..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 CANCEL sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>
;tag=18853e82KDe7j. .To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 CANCEL..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: 0.... #
U 67.33.160.119:18294 -> 10.194.206.102:5060 ACK sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524 431af92cef56-1--d87543-;rport..To: "45517705678570" < sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>
;tag=BXgB1FZBUZ3Da..F rom: "4000002"<sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm Y...CSeq: 2 ACK..Content-Length: 0.... #
U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From: "4000002" <s ip:0014444295793@184.72.206.204 <ip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Contact: s ip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL ..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA GE..Content-Length: 0.... # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..Fr om: "4000002" <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic. com>..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OP TIONS,INFO,MESSAGE..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 ACK sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4b KBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>
;tag=18853e82KDe7j..To : <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A CK..Content-Length: 0.... #
On Wed, Jul 7, 2010 at 5:50 PM, paul gore <paul.gore.j@gmail.com> wrote:
Seems like siptraffic uses 6 ip addresses for media, can that be the problem? Is there any setting in a gateway config xml which helps with that? I will do ngrep thing and update.
On 7/7/10, paul gore <paul.gore.j@gmail.com> wrote:
This provider does work on another box which is not natted as ec2. Most puzzling here though is why call originaion via api even not going via siptraffic still gets no audio.
On 7/7/10, Tony Graziano <tgraziano@myitdepartment.net> wrote:
You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo).
On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
What "doesn't work" means? It could be (and most likely is not) FS-related problem
On Wednesday 07 July 2010, Madovsky wrote:
I had same problem from this provider without to explain why. One day it works, another day it doesn't, their support is crap...
----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 07, 2010 2:37 PM Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
not really, not with so little information.
On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com> wrote:
Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one?
On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird > audio issue - I followed FreeSwitch EC2 wiki article and modified > internal profile > and vars.xml accordingly, but unfortunately still cannot get it > working. Incoming and outgoing calls made using a SIP phone to FS > extensions work just fine. As well as calls to FS from PSTN. But > calls to PSTN via gateways result in no audio at all, no ring, > nothing, SIP signaling goes through OK. Sofia status profile shows > correct values for Ext-RTP-IP for both profiles - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration > may be? > Please help, I really need to get this going. > > Thanks.
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Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
this sip trace is less than useful can you get a pcap file and put it where someone can wget it? /b On Jul 8, 2010, at 6:31 PM, paul gore wrote:
I got ngrep trace for port 5060 while making a call to a US number via siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I heard no audio not even ringing. Is there anything in this trace which can help identify the problem?
10.194.206.102:5060 - is my local EC2 IP 184.72.206.204:5060 - is my public EC2 IP 77.72.169.128:5060 - siptraffic.com proxy IP
Thanks!
I ever met same issue in EC2. Modify the sip_profile configuration file like "internal.xml" Replace the below item with actual ip address: <param name="ext-rtp-ip" value="xxx.xxx.xxx.xxx"/> Thanks, Will On Fri, Jul 9, 2010 at 7:31 AM, paul gore <paul.gore.j@gmail.com> wrote:
I got ngrep trace for port 5060 while making a call to a US number via siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I heard no audio not even ringing. Is there anything in this trace which can help identify the problem?
10.194.206.102:5060 - is my local EC2 IP 184.72.206.204:5060 - is my public EC2 IP 77.72.169.128:5060 - siptraffic.com proxy IP
Thanks!
67.33.160.119:18294 -> 10.194.206.102:5060 INVITE sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f 524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: < sip:4000002@67.33.160.119:18027>..To: "45517 709248570"<sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>..From: "4000002"<sip:4000002@myserver.com <sip%3A4000002@myserver.com>
;tag=5f1ec15f..Call- ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest user name="4000002",realm="myserver.com ",nonce="cf9019cc-f44a-4568-97d1-e9883fb1821f",uri="sip:45517705678570@v ersafon.com ",response="57da72527524e0e065c9a3221bfadd38",cnonce="140f655ff3427f6ba3767ab7040231f3",nc=0000 0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41150..Content-Length: 417....v=0..o=- 8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107 119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8 46298..a=fmtp:101 0-15..a=rtpmap:107 BV32/16000..a=rtpmap:119 BV32-FEC/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap :105 SPEEX-FEC/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r port=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To: "455177092 48570" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I NVITE..User-Agent: myserver..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 INVITE sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204> ;tag=18853e82KDe7j. .To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 INVITE..Contact: <sip:gw+voicetrading.com@184.72.206.204:5080 ;transport=udp;gw=voicetrading.com>..User-A gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.. Supported: timer, precondition, path, replaces..Allow-Events: talk, refer..Content-Type: application/sdp.. Content-Disposition: session..Content-Length: 295..X-FS-Support: update_display..Remote-Party-ID: "4000002 " <sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1278518039 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/80 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20.. # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..From : "4000002" <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.co<sip%3A0017705678570@sip.siptraffic.co> m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 198....v=0..o=C ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4 77.72.168.40..t=0 0..m=audio 57672 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20.. # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1- -d87543-;rport=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To: "45517705678570" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjk zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact: <sip:45517705678570@184.72.206.204:5060;transport=udp>..User-A gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message- summary, refer..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 251..Remote-P arty-ID: "45517705678570" <sip:45517705678570@10.194.206.102<sip%3A45517705678570@10.194.206.102> ;party=calling;privacy=off;screen=no....v=0.. o=FreeSWITCH 1278530815 1278530816 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m= audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=sil enceSupp:off - - - -..a=ptime:20.. #
U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..From : "4000002" <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.co<sip%3A0017705678570@sip.siptraffic.co> m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 204....v=0..o=C ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN IP4 208.167.230.118..t=0 0..m=audio 57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20.. #
U 67.33.160.119:18294 -> 10.194.206.102:5060 .... # U 67.33.160.119:18294 -> 10.194.206.102:5060 CANCEL sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f 524431af92cef56-1--d87543-;rport..To: "45517705678570"< sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>>..From: "4000002"<s ip:4000002@myserver.com <ip%3A4000002@myserver.com>>;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC EL..Proxy-Authorization: Digest username="4000002",realm="myserver.com ",nonce="cf9019cc-f44a-4568-97d1-e98 83fb1821f",uri="sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com> ",response="46c7e289f7490c807565c561699b03d6",cnonce="a226c
55446c605ee229f045602b29135",nc=00000002,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41 150..Content-Length: 0.... # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport =18294;received=67.33.160.119..From: "4000002" <sip:4000002@myserver.com<sip%3A4000002@myserver.com>>;tag=5f1ec15f..To: "4551770567857 0" <sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY mY...CSeq: 2 CANCEL..Content-Length: 0.... # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56- 1--d87543-;rport=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To : "45517705678570" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmN jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSA GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, repla ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presen ce.winfo, message-summary, refer..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 CANCEL sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>
;tag=18853e82KDe7j. .To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 CANCEL..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: 0.... #
U 67.33.160.119:18294 -> 10.194.206.102:5060 ACK sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524 431af92cef56-1--d87543-;rport..To: "45517705678570" < sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>
;tag=BXgB1FZBUZ3Da..F rom: "4000002"<sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm Y...CSeq: 2 ACK..Content-Length: 0.... #
U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From: "4000002" <s ip:0014444295793@184.72.206.204 <ip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Contact: s ip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL ..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA GE..Content-Length: 0.... # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..Fr om: "4000002" <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic. com>..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OP TIONS,INFO,MESSAGE..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 ACK sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4b KBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>
;tag=18853e82KDe7j..To : <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A CK..Content-Length: 0.... #
On Wed, Jul 7, 2010 at 5:50 PM, paul gore <paul.gore.j@gmail.com> wrote:
Seems like siptraffic uses 6 ip addresses for media, can that be the problem? Is there any setting in a gateway config xml which helps with that? I will do ngrep thing and update.
On 7/7/10, paul gore <paul.gore.j@gmail.com> wrote:
This provider does work on another box which is not natted as ec2. Most puzzling here though is why call originaion via api even not going via siptraffic still gets no audio.
On 7/7/10, Tony Graziano <tgraziano@myitdepartment.net> wrote:
You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo).
On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
What "doesn't work" means? It could be (and most likely is not) FS-related problem
On Wednesday 07 July 2010, Madovsky wrote:
I had same problem from this provider without to explain why. One day it works, another day it doesn't, their support is crap...
----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 07, 2010 2:37 PM Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
not really, not with so little information.
On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com> wrote:
Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one?
On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird > audio issue - I followed FreeSwitch EC2 wiki article and modified > internal profile > and vars.xml accordingly, but unfortunately still cannot get it > working. Incoming and outgoing calls made using a SIP phone to FS > extensions work just fine. As well as calls to FS from PSTN. But > calls to PSTN via gateways result in no audio at all, no ring, > nothing, SIP signaling goes through OK. Sofia status profile shows > correct values for Ext-RTP-IP for both profiles - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration > may be? > Please help, I really need to get this going. > > Thanks.
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You have provided only SIP log. Where is the media analysis? You should have run another ngrep to see if media traffic is OK between FS and siptraffic: "ngrep port 31564" I took the port value from the following line: c=IN IP4 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP
From the following SIP request:
U 10.194.206.102:5080 -> 77.72.169.128:5060 INVITE sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>
;tag=18853e82KDe7j. .To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 INVITE..Contact: <sip:gw+voicetrading.com@184.72.206.204:5080 ;transport=udp;gw=voicetrading.com>..User-A gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.. Supported: timer, precondition, path, replaces..Allow-Events: talk, refer..Content-Type: application/sdp.. Content-Disposition: session..Content-Length: 295..X-FS-Support: update_display..Remote-Party-ID: "4000002 " <sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1278518039 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/80 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20.. #
2010/7/9 k xd <kouxiaodong@gmail.com>
I ever met same issue in EC2.
Modify the sip_profile configuration file like "internal.xml" Replace the below item with actual ip address: <param name="ext-rtp-ip" value="xxx.xxx.xxx.xxx"/>
Thanks, Will
On Fri, Jul 9, 2010 at 7:31 AM, paul gore <paul.gore.j@gmail.com> wrote:
I got ngrep trace for port 5060 while making a call to a US number via siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I heard no audio not even ringing. Is there anything in this trace which can help identify the problem?
10.194.206.102:5060 - is my local EC2 IP 184.72.206.204:5060 - is my public EC2 IP 77.72.169.128:5060 - siptraffic.com proxy IP
Thanks!
67.33.160.119:18294 -> 10.194.206.102:5060 INVITE sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f 524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: < sip:4000002@67.33.160.119:18027>..To: "45517 709248570"<sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>..From: "4000002"<sip:4000002@myserver.com <sip%3A4000002@myserver.com>
;tag=5f1ec15f..Call- ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest user name="4000002",realm="myserver.com ",nonce="cf9019cc-f44a-4568-97d1-e9883fb1821f",uri="sip:45517705678570@v ersafon.com ",response="57da72527524e0e065c9a3221bfadd38",cnonce="140f655ff3427f6ba3767ab7040231f3",nc=0000 0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41150..Content-Length: 417....v=0..o=- 8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107 119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8 46298..a=fmtp:101 0-15..a=rtpmap:107 BV32/16000..a=rtpmap:119 BV32-FEC/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap :105 SPEEX-FEC/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r port=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To: "455177092 48570" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I NVITE..User-Agent: myserver..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 INVITE sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204> ;tag=18853e82KDe7j. .To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 INVITE..Contact: <sip:gw+voicetrading.com@184.72.206.204:5080 ;transport=udp;gw=voicetrading.com>..User-A gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.. Supported: timer, precondition, path, replaces..Allow-Events: talk, refer..Content-Type: application/sdp.. Content-Disposition: session..Content-Length: 295..X-FS-Support: update_display..Remote-Party-ID: "4000002 " <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1278518039 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/80 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20.. # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..From : "4000002" <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.co<sip%3A0017705678570@sip.siptraffic.co> m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 198....v=0..o=C ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4 77.72.168.40..t=0 0..m=audio 57672 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20.. # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1- -d87543-;rport=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To: "45517705678570" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjk zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact: <sip:45517705678570@184.72.206.204:5060;transport=udp>..User-A gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message- summary, refer..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 251..Remote-P arty-ID: "45517705678570" <sip:45517705678570@10.194.206.102<sip%3A45517705678570@10.194.206.102> ;party=calling;privacy=off;screen=no....v=0.. o=FreeSWITCH 1278530815 1278530816 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m= audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=sil enceSupp:off - - - -..a=ptime:20.. #
U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..From : "4000002" <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.co<sip%3A0017705678570@sip.siptraffic.co> m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 204....v=0..o=C ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN IP4 208.167.230.118..t=0 0..m=audio 57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20.. #
U 67.33.160.119:18294 -> 10.194.206.102:5060 .... # U 67.33.160.119:18294 -> 10.194.206.102:5060 CANCEL sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f 524431af92cef56-1--d87543-;rport..To: "45517705678570"< sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>>..From: "4000002"<s ip:4000002@myserver.com <ip%3A4000002@myserver.com>>;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC EL..Proxy-Authorization: Digest username="4000002",realm="myserver.com ",nonce="cf9019cc-f44a-4568-97d1-e98 83fb1821f",uri="sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com> ",response="46c7e289f7490c807565c561699b03d6",cnonce="a226c
55446c605ee229f045602b29135",nc=00000002,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41 150..Content-Length: 0.... # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport =18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To: "4551770567857 0" <sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY mY...CSeq: 2 CANCEL..Content-Length: 0.... # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56- 1--d87543-;rport=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To : "45517705678570" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmN jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSA GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, repla ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presen ce.winfo, message-summary, refer..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 CANCEL sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>
;tag=18853e82KDe7j. .To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 CANCEL..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: 0.... #
U 67.33.160.119:18294 -> 10.194.206.102:5060 ACK sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524 431af92cef56-1--d87543-;rport..To: "45517705678570" < sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>
;tag=BXgB1FZBUZ3Da..F rom: "4000002"<sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm Y...CSeq: 2 ACK..Content-Length: 0.... #
U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From: "4000002" <s ip:0014444295793@184.72.206.204 <ip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Contact: s ip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL ..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA GE..Content-Length: 0.... # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..Fr om: "4000002" <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic. com>..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OP TIONS,INFO,MESSAGE..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 ACK sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4b KBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>
;tag=18853e82KDe7j..To : <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A CK..Content-Length: 0.... #
On Wed, Jul 7, 2010 at 5:50 PM, paul gore <paul.gore.j@gmail.com> wrote:
Seems like siptraffic uses 6 ip addresses for media, can that be the problem? Is there any setting in a gateway config xml which helps with that? I will do ngrep thing and update.
On 7/7/10, paul gore <paul.gore.j@gmail.com> wrote:
This provider does work on another box which is not natted as ec2. Most puzzling here though is why call originaion via api even not going via siptraffic still gets no audio.
On 7/7/10, Tony Graziano <tgraziano@myitdepartment.net> wrote:
You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo).
On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
What "doesn't work" means? It could be (and most likely is not) FS-related problem
On Wednesday 07 July 2010, Madovsky wrote: > I had same problem from this provider without to explain why. > One day it works, another day it doesn't, their support is crap... > > ----- Original Message ----- > From: Anthony Minessale > To: freeswitch-users@lists.freeswitch.org > Sent: Wednesday, July 07, 2010 2:37 PM > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on > outgoing > calls > > > not really, not with so little information. > > > > On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com
wrote: > > Firewall is configured according to the wiki, I also tried to open all > udp ports, issue persists. > Actually the problem became more complex - outgoing calls don't > work > with one particular termination provider, siptraffic.com , any > ideas > why? > Outgoing calls also don't work when originating a call via js > script > or via FS api. Any clues on that one? > > On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > > Hi there, > > I am experimenting with FS on EC2, I like results, but stuck on weird > > audio issue - I followed FreeSwitch EC2 wiki article and > modified > > internal profile > > and vars.xml accordingly, but unfortunately still cannot get it > > working. Incoming and outgoing calls made using a SIP phone to > FS > > extensions work just fine. As well as calls to FS from PSTN. But > > calls to PSTN via gateways result in no audio at all, no ring, > > nothing, SIP signaling goes through OK. Sofia status profile > shows > > correct values for Ext-RTP-IP for both profiles - > > my static public IP, RTP-IP shows local IP. > > Any thoughts on that? Anybody can share working profile configuration > > may be? > > Please help, I really need to get this going. > > > > Thanks.
> > _______________________________________________ > 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 > > > > > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > > MSN:anthony_minessale@hotmail.com<MSN%3Aanthony_minessale@hotmail.com> <MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> <PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
> IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> <sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
> > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> <googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
> pstn:+19193869900 > > > >
> ---
> > > _______________________________________________ > 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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-- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427
Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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You also need to set ext-sip-ip and ext-rtp-ip also can you get me a pcap? I know it works because I have personally done this on EC2 along with many others. Its as simple as understanding NAT and FreeSWITCH. /b On Jul 9, 2010, at 2:57 AM, Vitalii Colosov wrote:
You should have run another ngrep to see if media traffic is OK between FS and siptraffic:
I second that, I have one FS running on EC2 instance for production, so far no complaints yet. You do need good understanding of both NAT and FreeSWITCH. Chris On Fri, Jul 9, 2010 at 9:35 AM, Brian West <brian@freeswitch.org> wrote:
You also need to set ext-sip-ip and ext-rtp-ip also can you get me a pcap? I know it works because I have personally done this on EC2 along with many others.
Its as simple as understanding NAT and FreeSWITCH.
/b
On Jul 9, 2010, at 2:57 AM, Vitalii Colosov wrote:
You should have run another ngrep to see if media traffic is OK between FS and siptraffic:
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As I mentioned in my first post I did that, I did everything as per ec2 wiki, incuding setting ext-ip in profiles. I have everything working fine, except no audio via siptraffic.com. I will do pcap and rtp trace via ngrep over weekend and update. On 7/9/10, k xd <kouxiaodong@gmail.com> wrote:
I ever met same issue in EC2.
Modify the sip_profile configuration file like "internal.xml" Replace the below item with actual ip address: <param name="ext-rtp-ip" value="xxx.xxx.xxx.xxx"/>
Thanks, Will
On Fri, Jul 9, 2010 at 7:31 AM, paul gore <paul.gore.j@gmail.com> wrote:
I got ngrep trace for port 5060 while making a call to a US number via siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I heard no audio not even ringing. Is there anything in this trace which can help identify the problem?
10.194.206.102:5060 - is my local EC2 IP 184.72.206.204:5060 - is my public EC2 IP 77.72.169.128:5060 - siptraffic.com proxy IP
Thanks!
67.33.160.119:18294 -> 10.194.206.102:5060 INVITE sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f 524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: < sip:4000002@67.33.160.119:18027>..To: "45517
709248570"<sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>..From: "4000002"<sip:4000002@myserver.com <sip%3A4000002@myserver.com>
;tag=5f1ec15f..Call- ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest user name="4000002",realm="myserver.com ",nonce="cf9019cc-f44a-4568-97d1-e9883fb1821f",uri="sip:45517705678570@v ersafon.com ",response="57da72527524e0e065c9a3221bfadd38",cnonce="140f655ff3427f6ba3767ab7040231f3",nc=0000 0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41150..Content-Length: 417....v=0..o=- 8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107 119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8 46298..a=fmtp:101 0-15..a=rtpmap:107 BV32/16000..a=rtpmap:119 BV32-FEC/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap :105 SPEEX-FEC/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r port=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To: "455177092 48570" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I NVITE..User-Agent: myserver..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 INVITE sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204> ;tag=18853e82KDe7j. .To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 INVITE..Contact: <sip:gw+voicetrading.com@184.72.206.204:5080 ;transport=udp;gw=voicetrading.com>..User-A gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.. Supported: timer, precondition, path, replaces..Allow-Events: talk, refer..Content-Type: application/sdp.. Content-Disposition: session..Content-Length: 295..X-FS-Support: update_display..Remote-Party-ID: "4000002 " <sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1278518039 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/80 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20.. # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..From : "4000002" <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.co<sip%3A0017705678570@sip.siptraffic.co> m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 198....v=0..o=C ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4 77.72.168.40..t=0 0..m=audio 57672 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20.. # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1- -d87543-;rport=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To: "45517705678570" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjk zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact: <sip:45517705678570@184.72.206.204:5060;transport=udp>..User-A gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message- summary, refer..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 251..Remote-P arty-ID: "45517705678570" <sip:45517705678570@10.194.206.102<sip%3A45517705678570@10.194.206.102> ;party=calling;privacy=off;screen=no....v=0.. o=FreeSWITCH 1278530815 1278530816 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m= audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=sil enceSupp:off - - - -..a=ptime:20.. #
U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..From : "4000002" <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.co<sip%3A0017705678570@sip.siptraffic.co> m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 204....v=0..o=C ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN IP4 208.167.230.118..t=0 0..m=audio 57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20.. #
U 67.33.160.119:18294 -> 10.194.206.102:5060 .... # U 67.33.160.119:18294 -> 10.194.206.102:5060 CANCEL sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f 524431af92cef56-1--d87543-;rport..To: "45517705678570"< sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>>..From: "4000002"<s ip:4000002@myserver.com <ip%3A4000002@myserver.com>>;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC EL..Proxy-Authorization: Digest username="4000002",realm="myserver.com ",nonce="cf9019cc-f44a-4568-97d1-e98
83fb1821f",uri="sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com> ",response="46c7e289f7490c807565c561699b03d6",cnonce="a226c
55446c605ee229f045602b29135",nc=00000002,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41 150..Content-Length: 0.... # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport =18294;received=67.33.160.119..From: "4000002" <sip:4000002@myserver.com<sip%3A4000002@myserver.com>>;tag=5f1ec15f..To: "4551770567857 0" <sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY mY...CSeq: 2 CANCEL..Content-Length: 0.... # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56- 1--d87543-;rport=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..To : "45517705678570" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmN jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSA GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, repla ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presen ce.winfo, message-summary, refer..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 CANCEL sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>
;tag=18853e82KDe7j. .To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 CANCEL..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: 0.... #
U 67.33.160.119:18294 -> 10.194.206.102:5060 ACK sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524 431af92cef56-1--d87543-;rport..To: "45517705678570" < sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>
;tag=BXgB1FZBUZ3Da..F rom: "4000002"<sip:4000002@myserver.com <sip%3A4000002@myserver.com>>;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm Y...CSeq: 2 ACK..Content-Length: 0.... #
U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From: "4000002" <s ip:0014444295793@184.72.206.204 <ip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Contact: s ip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL ..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA GE..Content-Length: 0.... # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..Fr om: "4000002" <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204>>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic. com>..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OP TIONS,INFO,MESSAGE..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 ACK sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4b KBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>
;tag=18853e82KDe7j..To : <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com>>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A CK..Content-Length: 0.... #
On Wed, Jul 7, 2010 at 5:50 PM, paul gore <paul.gore.j@gmail.com> wrote:
Seems like siptraffic uses 6 ip addresses for media, can that be the problem? Is there any setting in a gateway config xml which helps with that? I will do ngrep thing and update.
On 7/7/10, paul gore <paul.gore.j@gmail.com> wrote:
This provider does work on another box which is not natted as ec2. Most puzzling here though is why call originaion via api even not going via siptraffic still gets no audio.
On 7/7/10, Tony Graziano <tgraziano@myitdepartment.net> wrote:
You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo).
On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
What "doesn't work" means? It could be (and most likely is not) FS-related problem
On Wednesday 07 July 2010, Madovsky wrote: > I had same problem from this provider without to explain why. > One day it works, another day it doesn't, their support is crap... > > ----- Original Message ----- > From: Anthony Minessale > To: freeswitch-users@lists.freeswitch.org > Sent: Wednesday, July 07, 2010 2:37 PM > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on > outgoing > calls > > > not really, not with so little information. > > > > On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com> wrote: > > Firewall is configured according to the wiki, I also tried to open all > udp ports, issue persists. > Actually the problem became more complex - outgoing calls don't > work > with one particular termination provider, siptraffic.com , any > ideas > why? > Outgoing calls also don't work when originating a call via js > script > or via FS api. Any clues on that one? > > On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > > Hi there, > > I am experimenting with FS on EC2, I like results, but stuck on weird > > audio issue - I followed FreeSwitch EC2 wiki article and > modified > > internal profile > > and vars.xml accordingly, but unfortunately still cannot get it > > working. Incoming and outgoing calls made using a SIP phone > to > FS > > extensions work just fine. As well as calls to FS from PSTN. But > > calls to PSTN via gateways result in no audio at all, no > ring, > > nothing, SIP signaling goes through OK. Sofia status profile > shows > > correct values for Ext-RTP-IP for both profiles - > > my static public IP, RTP-IP shows local IP. > > Any thoughts on that? Anybody can share working profile configuration > > may be? > > Please help, I really need to get this going. > > > > Thanks.
> > _______________________________________________ > 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 > > > > > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > > MSN:anthony_minessale@hotmail.com<MSN%3Aanthony_minessale@hotmail.com> <MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> <PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
> IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> <sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
> > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> <googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
> pstn:+19193869900 > > > >
> ---
> > > _______________________________________________ > 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 >
_______________________________________________ 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
-- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427
Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
_______________________________________________ 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
in the future post logs to pastebin, notice how big and ugly this thread is now quoting your log and toting it around on each reply. Also consider coming to IRC for real-time interaction and post the conclusion here once you figure it out, On Fri, Jul 9, 2010 at 9:20 AM, paul gore <paul.gore.j@gmail.com> wrote:
As I mentioned in my first post I did that, I did everything as per ec2 wiki, incuding setting ext-ip in profiles. I have everything working fine, except no audio via siptraffic.com. I will do pcap and rtp trace via ngrep over weekend and update.
On 7/9/10, k xd <kouxiaodong@gmail.com> wrote:
I ever met same issue in EC2.
Modify the sip_profile configuration file like "internal.xml" Replace the below item with actual ip address: <param name="ext-rtp-ip" value="xxx.xxx.xxx.xxx"/>
Thanks, Will
On Fri, Jul 9, 2010 at 7:31 AM, paul gore <paul.gore.j@gmail.com> wrote:
I got ngrep trace for port 5060 while making a call to a US number via siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I heard no audio not even ringing. Is there anything in this trace which can help identify the problem?
10.194.206.102:5060 - is my local EC2 IP 184.72.206.204:5060 - is my public EC2 IP 77.72.169.128:5060 - siptraffic.com proxy IP
Thanks!
67.33.160.119:18294 -> 10.194.206.102:5060 INVITE sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>< sip%3A45517705678570@myserver.com <sip%253A45517705678570@myserver.com> SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f 524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: < sip:4000002@67.33.160.119:18027>..To: "45517
709248570"<sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com> <sip%3A45517705678570@myserver.com <sip%253A45517705678570@myserver.com> ..From: "4000002"<sip:4000002@myserver.com <sip%3A4000002@myserver.com> < sip%3A4000002@myserver.com <sip%253A4000002@myserver.com>>
;tag=5f1ec15f..Call- ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest user name="4000002",realm="myserver.com ",nonce="cf9019cc-f44a-4568-97d1-e9883fb1821f",uri="sip:45517705678570@v ersafon.com
",response="57da72527524e0e065c9a3221bfadd38",cnonce="140f655ff3427f6ba3767ab7040231f3",nc=0000
0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41150..Content-Length: 417....v=0..o=- 8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107 119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8 46298..a=fmtp:101 0-15..a=rtpmap:107 BV32/16000..a=rtpmap:119 BV32-FEC/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap :105 SPEEX-FEC/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r port=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com> < sip%3A4000002@myserver.com <sip%253A4000002@myserver.com> ;tag=5f1ec15f..To: "455177092 48570" <sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>< sip%3A45517705678570@myserver.com <sip%253A45517705678570@myserver.com> ..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I NVITE..User-Agent: myserver..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 INVITE sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com> <sip%3A0017705678570@sip.siptraffic.com<sip%253A0017705678570@sip.siptraffic.com> SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204> < sip%3A0014444295793@184.72.206.204 <sip%253A0014444295793@184.72.206.204>>
;tag=18853e82KDe7j. .To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com> <sip%3A0017705678570@sip.siptraffic.com<sip%253A0017705678570@sip.siptraffic.com> ..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 INVITE..Contact: <sip:gw+voicetrading.com@184.72.206.204:5080 ;transport=udp;gw=voicetrading.com>..User-A gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.. Supported: timer, precondition, path, replaces..Allow-Events: talk, refer..Content-Type: application/sdp.. Content-Disposition: session..Content-Length: 295..X-FS-Support: update_display..Remote-Party-ID: "4000002 " <sip:0014444295793@184.72.206.204<sip%3A0014444295793@184.72.206.204> <sip%3A0014444295793@184.72.206.204<sip%253A0014444295793@184.72.206.204> ;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1278518039 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/80 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20.. # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..From : "4000002" <sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>< sip%3A0014444295793@184.72.206.204 <sip%253A0014444295793@184.72.206.204> ;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.co<sip%3A0017705678570@sip.siptraffic.co> <sip%3A0017705678570@sip.siptraffic.co<sip%253A0017705678570@sip.siptraffic.co>
m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 198....v=0..o=C ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4 77.72.168.40..t=0 0..m=audio 57672 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20.. # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1- -d87543-;rport=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com> < sip%3A4000002@myserver.com <sip%253A4000002@myserver.com> ;tag=5f1ec15f..To: "45517705678570" <sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>< sip%3A45517705678570@myserver.com <sip%253A45517705678570@myserver.com> ;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjk zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact: <sip:45517705678570@184.72.206.204:5060;transport=udp>..User-A gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message- summary, refer..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 251..Remote-P arty-ID: "45517705678570" <sip:45517705678570@10.194.206.102<sip%3A45517705678570@10.194.206.102> <sip%3A45517705678570@10.194.206.102<sip%253A45517705678570@10.194.206.102>
;party=calling;privacy=off;screen=no....v=0.. o=FreeSWITCH 1278530815 1278530816 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m= audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=sil enceSupp:off - - - -..a=ptime:20.. #
U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..From : "4000002" <sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>< sip%3A0014444295793@184.72.206.204 <sip%253A0014444295793@184.72.206.204> ;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.co<sip%3A0017705678570@sip.siptraffic.co> <sip%3A0017705678570@sip.siptraffic.co<sip%253A0017705678570@sip.siptraffic.co>
m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 204....v=0..o=C ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN IP4 208.167.230.118..t=0 0..m=audio 57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20.. #
U 67.33.160.119:18294 -> 10.194.206.102:5060 .... # U 67.33.160.119:18294 -> 10.194.206.102:5060 CANCEL sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>< sip%3A45517705678570@myserver.com <sip%253A45517705678570@myserver.com> SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f 524431af92cef56-1--d87543-;rport..To: "45517705678570"< sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com> <sip%3A45517705678570@myserver.com<sip%253A45517705678570@myserver.com> ..From: "4000002"<s ip:4000002@myserver.com <ip%3A4000002@myserver.com> <ip%3A4000002@myserver.com <ip%253A4000002@myserver.com> ;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC EL..Proxy-Authorization: Digest username="4000002",realm=" myserver.com ",nonce="cf9019cc-f44a-4568-97d1-e98
83fb1821f",uri="sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com> <sip%3A45517705678570@myserver.com <sip%253A45517705678570@myserver.com>> ",response="46c7e289f7490c807565c561699b03d6",cnonce="a226c
55446c605ee229f045602b29135",nc=00000002,qop=auth,algorithm=MD5..User-Agent:
X-Lite release 1011s stamp 41 150..Content-Length: 0.... # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport =18294;received=67.33.160.119..From: "4000002" <sip:4000002@myserver.com <sip%3A4000002@myserver.com>< sip%3A4000002@myserver.com <sip%253A4000002@myserver.com> ;tag=5f1ec15f..To: "4551770567857 0" <sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com> <sip%3A45517705678570@myserver.com<sip%253A45517705678570@myserver.com> ;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY mY...CSeq: 2 CANCEL..Content-Length: 0.... # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56- 1--d87543-;rport=18294;received=67.33.160.119..From: "4000002" < sip:4000002@myserver.com <sip%3A4000002@myserver.com> < sip%3A4000002@myserver.com <sip%253A4000002@myserver.com> ;tag=5f1ec15f..To : "45517705678570" <sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com>< sip%3A45517705678570@myserver.com <sip%253A45517705678570@myserver.com> ;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmN jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSA GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, repla ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presen ce.winfo, message-summary, refer..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 CANCEL sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com> <sip%3A0017705678570@sip.siptraffic.com<sip%253A0017705678570@sip.siptraffic.com> SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204> < sip%3A0014444295793@184.72.206.204 <sip%253A0014444295793@184.72.206.204>>
;tag=18853e82KDe7j. .To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com> <sip%3A0017705678570@sip.siptraffic.com<sip%253A0017705678570@sip.siptraffic.com> ..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 CANCEL..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: 0.... #
U 67.33.160.119:18294 -> 10.194.206.102:5060 ACK sip:45517705678570@myserver.com<sip%3A45517705678570@myserver.com> <sip%3A45517705678570@myserver.com<sip%253A45517705678570@myserver.com>>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524 431af92cef56-1--d87543-;rport..To: "45517705678570" < sip:45517705678570@myserver.com <sip%3A45517705678570@myserver.com> < sip%3A45517705678570@myserver.com <sip%253A45517705678570@myserver.com>>
;tag=BXgB1FZBUZ3Da..F rom: "4000002"<sip:4000002@myserver.com <sip%3A4000002@myserver.com> <sip%3A4000002@myserver.com <sip%253A4000002@myserver.com> ;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm Y...CSeq: 2 ACK..Content-Length: 0.... #
U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From: "4000002" <s ip:0014444295793@184.72.206.204 <ip%3A0014444295793@184.72.206.204> <ip%3A0014444295793@184.72.206.204<ip%253A0014444295793@184.72.206.204> ;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com> <sip%3A0017705678570@sip.siptraffic.com<sip%253A0017705678570@sip.siptraffic.com> ..Contact: s ip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL ..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA GE..Content-Length: 0.... # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..Fr om: "4000002" <sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204>< sip%3A0014444295793@184.72.206.204 <sip%253A0014444295793@184.72.206.204> ;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic. com>..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OP TIONS,INFO,MESSAGE..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 ACK sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com> <sip%3A0017705678570@sip.siptraffic.com<sip%253A0017705678570@sip.siptraffic.com> SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4b KBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793@184.72.206.204 <sip%3A0014444295793@184.72.206.204> < sip%3A0014444295793@184.72.206.204 <sip%253A0014444295793@184.72.206.204>>
;tag=18853e82KDe7j..To : <sip:0017705678570@sip.siptraffic.com<sip%3A0017705678570@sip.siptraffic.com> <sip%3A0017705678570@sip.siptraffic.com<sip%253A0017705678570@sip.siptraffic.com> ..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A CK..Content-Length: 0.... #
On Wed, Jul 7, 2010 at 5:50 PM, paul gore <paul.gore.j@gmail.com> wrote:
Seems like siptraffic uses 6 ip addresses for media, can that be the problem? Is there any setting in a gateway config xml which helps with that? I will do ngrep thing and update.
On 7/7/10, paul gore <paul.gore.j@gmail.com> wrote:
This provider does work on another box which is not natted as ec2. Most puzzling here though is why call originaion via api even not going via siptraffic still gets no audio.
On 7/7/10, Tony Graziano <tgraziano@myitdepartment.net> wrote:
You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo).
On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
> What "doesn't work" means? It could be (and most likely is not) > FS-related > problem > > On Wednesday 07 July 2010, Madovsky wrote: > > I had same problem from this provider without to explain why. > > One day it works, another day it doesn't, their support is crap... > > > > ----- Original Message ----- > > From: Anthony Minessale > > To: freeswitch-users@lists.freeswitch.org > > Sent: Wednesday, July 07, 2010 2:37 PM > > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on > > outgoing > > calls > > > > > > not really, not with so little information. > > > > > > > > On Wed, Jul 7, 2010 at 1:30 PM, paul gore < paul.gore.j@gmail.com> > wrote: > > > > Firewall is configured according to the wiki, I also tried to open > all > > udp ports, issue persists. > > Actually the problem became more complex - outgoing calls don't > > work > > with one particular termination provider, siptraffic.com , any > > ideas > > why? > > Outgoing calls also don't work when originating a call via js > > script > > or via FS api. Any clues on that one? > > > > On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > > > Hi there, > > > I am experimenting with FS on EC2, I like results, but stuck on > weird > > > audio issue - I followed FreeSwitch EC2 wiki article and > > modified > > > internal profile > > > and vars.xml accordingly, but unfortunately still cannot get it > > > working. Incoming and outgoing calls made using a SIP phone > > to > > FS > > > extensions work just fine. As well as calls to FS from PSTN. But > > > calls to PSTN via gateways result in no audio at all, no > > ring, > > > nothing, SIP signaling goes through OK. Sofia status profile > > shows > > > correct values for Ext-RTP-IP for both profiles - > > > my static public IP, RTP-IP shows local IP. > > > Any thoughts on that? Anybody can share working profile > configuration > > > may be? > > > Please help, I really need to get this going. > > > > > > Thanks.
> > > > _______________________________________________ > > 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 > > > > > > > > > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > > > MSN:anthony_minessale@hotmail.com<MSN%3Aanthony_minessale@hotmail.com> <MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com>
<MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com> <MSN%253Aanthony_minessale@hotmail.com<MSN%25253Aanthony_minessale@hotmail.com>
> > > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com>
<PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
<PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com> <PAYPAL%253Aanthony.minessale@gmail.com<PAYPAL%25253Aanthony.minessale@gmail.com>
> > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > > > sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org>
<sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
<sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org> <sip%253A888@conference.freeswitch.org<sip%25253A888@conference.freeswitch.org>
> > > > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org>
<googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
<googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org> <googletalk%253Aconf%252B888@conference.freeswitch.org<googletalk%25253Aconf%25252B888@conference.freeswitch.org>
> > pstn:+19193869900 > > > > > > > > >
> > ---
> > > > > > _______________________________________________ > > 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 > > > > > _______________________________________________ > 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 >
-- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427
Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Regarding the following: "Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one?" What do you see in the log? CALL_REJECTED? If so, try adding your FS IP address to the list of allowed IPs on their site - looks like they changed something and you can't use (until they fix) only your login and password. If "don't work" means something else, then too less info. :-( Maybe you could provide some logs of the call attempts... Regards, Vitalie 2010/7/7 Tony Graziano <tgraziano@myitdepartment.net>
You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo).
On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
What "doesn't work" means? It could be (and most likely is not) FS-related problem
On Wednesday 07 July 2010, Madovsky wrote:
I had same problem from this provider without to explain why. One day it works, another day it doesn't, their support is crap...
----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 07, 2010 2:37 PM Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
not really, not with so little information.
On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j@gmail.com> wrote:
Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one?
On 7/6/10, paul gore <paul.gore.j@gmail.com> wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird > audio issue - I followed FreeSwitch EC2 wiki article and modified > internal profile > and vars.xml accordingly, but unfortunately still cannot get it > working. Incoming and outgoing calls made using a SIP phone to FS > extensions work just fine. As well as calls to FS from PSTN. But > calls to PSTN via gateways result in no audio at all, no ring, > nothing, SIP signaling goes through OK. Sofia status profile shows > correct values for Ext-RTP-IP for both profiles - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration > may be? > Please help, I really need to get this going. > > Thanks.
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FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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-- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427
Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
_______________________________________________ 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
participants (9)
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Anthony Minessale
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Brian West
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Chris Chen
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k xd
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Madovsky
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paul gore
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Sergey Okhapkin
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Tony Graziano
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Vitalii Colosov