Finally an iPhone app that connects with SIP to FreeSWITCH. I tried it out, and it works perfect with my FreeSWITCH installation. Can be downloaded at iTunes App Store. Ivan
On 10/4/2008 4:47 AM, Ivan C Myrvold wrote:
Finally an iPhone app that connects with SIP to FreeSWITCH. I tried it out, and it works perfect with my FreeSWITCH installation. Can be downloaded at iTunes App Store.
Ivan
Ivan, I've been playing with fring. The only issue I have is that I can't get the registered fring client to hit anything but the 'public' dialplan context. I have: <variables> <variable name="user_context" value="default"/> </variables> set on the userid I'm registering fring with. Have you been able to get it to hit your default dialplan context?
On 4. okt.. 2008, at 14:14, Rupa Schomaker wrote:
On 10/4/2008 4:47 AM, Ivan C Myrvold wrote:
Finally an iPhone app that connects with SIP to FreeSWITCH. I tried it out, and it works perfect with my FreeSWITCH installation. Can be downloaded at iTunes App Store.
Ivan
Ivan, I've been playing with fring. The only issue I have is that I can't get the registered fring client to hit anything but the 'public' dialplan context.
I have:
<variables> <variable name="user_context" value="default"/> </variables>
set on the userid I'm registering fring with.
Have you been able to get it to hit your default dialplan context?
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Yes, after som more playing with it, I also have problems receiving calls. Originating is OK. I first thought this was an ordinary SIP client running on the iPhone, but I see now it isn't. Ivan
On Oct 4, 2008, at 8:39 AM, Ivan C Myrvold wrote:
Yes, after som more playing with it, I also have problems receiving calls. Originating is OK. I first thought this was an ordinary SIP client running on the iPhone, but I see now it isn't.
Like all iphone apps, its only running while its on the screen. If you put the phone to sleep or go back to the menu or any other app its going to not be able to receive a call. I have not tested this myself, but will do so today. MIke
On Oct 4, 2008, at 2:47 AM, Ivan C Myrvold wrote:
Finally an iPhone app that connects with SIP to FreeSWITCH. I tried it out, and it works perfect with my FreeSWITCH installation. Can be downloaded at iTunes App Store.
There is also Siphon which is available to jailbroken iphones. This appears tome to be more of a "real" SIP client... FYI, Marty
On Oct 4, 2008, at 9:31 AM, Brian West wrote:
Does this work on iphone firmware 2.x? Huh, I can't seem to find any info other then support for 1.1x firmware, so I don't know.
Didn't realize that. Thanks for the heads up, Marty
/b
On Oct 4, 2008, at 11:27 AM, Martin Joseph wrote:
There is also Siphon which is available to jailbroken iphones. This appears tome to be more of a "real" SIP client...
FYI, Marty
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Fring uses private address in contact.fields on some servers ... on reporting this I got this answer : Hello, Thank you for contacting our support team. I've checked with our SIP guy here and it seems like this is how other SIP softwares work (X-Lite for example) According to the SIP stack, Contact header MAY exist and it is not a MUST parameter. If you have further queries, please do not hesitate to contact us again. Thank you, Omri. fring support team. regards Torstein On Sat, Oct 4, 2008 at 7:00 PM, Martin Joseph <astmac@stillnewt.org> wrote:
On Oct 4, 2008, at 9:31 AM, Brian West wrote:
Does this work on iphone firmware 2.x? Huh, I can't seem to find any info other then support for 1.1x firmware, so I don't know.
Didn't realize that. Thanks for the heads up, Marty
/b
On Oct 4, 2008, at 11:27 AM, Martin Joseph wrote:
There is also Siphon which is available to jailbroken iphones. This appears tome to be more of a "real" SIP client...
FYI, Marty
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That should be framed so nobody can forget how they passed the buck to.... "the SIP stack" the all knowing SIP stack. What does that mean? Tell him next time he wants to pretend he knows what he's talking about to say "THE RFC SAYS...." That's how the pros do it. For example... The RFC says that if you call someone with a contact header that you MUST send any subsequent communication to that contact address so if you send FS an INVITE with a contact with a local addr, we will try to send the rest of the signaling to that addr. The solution is for them to use STUN, we all know they won't so the other solution is to turn on the NAT hacks in FS that will make us do an acl lookup on the contact addr and lock the traffic to be pumped out the reverse port. In your sip profile add the params <param name="apply_nat_acl" value="rfc918"/> <param name="agressive_nat_detection" value="true"/> This will make FS break all the normal rules so the phone can be behind nat and be stupid and blissful while we pretend they did the right thing and lie to them so they think you are calling somone on the local lan. On Wed, Oct 8, 2008 at 7:48 AM, Torstein Knutsen <torstein.knutsen@gmail.com
wrote:
Fring uses private address in contact.fields on some servers ... on reporting this I got this answer :
Hello, Thank you for contacting our support team.
I've checked with our SIP guy here and it seems like this is how other SIP softwares work (X-Lite for example) According to the SIP stack, Contact header MAY exist and it is not a MUST parameter.
If you have further queries, please do not hesitate to contact us again. Thank you, Omri. fring support team.
regards Torstein
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
In your sip profile add the params <param name="apply_nat_acl" value="rfc918"/> <param name="agressive_nat_detection" value="true"/> Just a quick question - is that a typo? Shouldn't it be "rfc1918" and not "rfc918"? I want to confirm this so that we can get it properly documented on the wiki. -MC
yep, sorry On Wed, Oct 8, 2008 at 11:06 AM, Michael Collins <mcollins@fcnetwork.com>wrote:
In your sip profile add the params <param name="apply_nat_acl" value="rfc918"/> <param name="agressive_nat_detection" value="true"/>
Just a quick question – is that a typo? Shouldn't it be "rfc1918" and not "rfc918"? I want to confirm this so that we can get it properly documented on the wiki.
-MC
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
It was a typo. Also the params have dashes between them... The standard we try to follow is params have dashes... and variables have underscores. /b On Oct 8, 2008, at 11:06 AM, Michael Collins wrote:
In your sip profile add the params <param name="apply_nat_acl" value="rfc918"/> <param name="agressive_nat_detection" value="true"/>
Just a quick question – is that a typo? Shouldn’t it be “rfc1918” and not “rfc918”? I want to confirm this so that we can get it properly documented on the wiki. -MC
Yeah, this is a good answer :) I did somemore research with Fring,and it seems that i'ts not the iphone that talks SIP, iphone talks "fring'ish" with fring servers, who in turn talks SIP with eg your FreeSWITCH .. I have yet to hear of somone having less than 1000ms delay on the media, so I think ("as the SIP guy over here") that it's better to hold out for another app to the iphone for SIP functionality... Some of these SIP servers uses official addresses in CONTACT, and some not. Servers I have seen has been located (if RIPE is correct) in Finland, Israel, and UK.. there probably is more ... anyhow tought you might like to know. regds Torstein On Wed, Oct 8, 2008 at 3:10 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
That should be framed so nobody can forget how they passed the buck to.... "the SIP stack" the all knowing SIP stack. What does that mean? Tell him next time he wants to pretend he knows what he's talking about to say "THE RFC SAYS...." That's how the pros do it.
For example... The RFC says that if you call someone with a contact header that you MUST send any subsequent communication to that contact address so if you send FS an INVITE with a contact with a local addr, we will try to send the rest of the signaling to that addr.
The solution is for them to use STUN, we all know they won't so the other solution is to turn on the NAT hacks in FS that will make us do an acl lookup on the contact addr and lock the traffic to be pumped out the reverse port.
In your sip profile add the params <param name="apply_nat_acl" value="rfc918"/> <param name="agressive_nat_detection" value="true"/>
This will make FS break all the normal rules so the phone can be behind nat and be stupid and blissful while we pretend they did the right thing and lie to them so they think you are calling somone on the local lan.
On Wed, Oct 8, 2008 at 7:48 AM, Torstein Knutsen < torstein.knutsen@gmail.com> wrote:
Fring uses private address in contact.fields on some servers ... on reporting this I got this answer :
Hello, Thank you for contacting our support team.
I've checked with our SIP guy here and it seems like this is how other SIP softwares work (X-Lite for example) According to the SIP stack, Contact header MAY exist and it is not a MUST parameter.
If you have further queries, please do not hesitate to contact us again. Thank you, Omri. fring support team.
regards Torstein
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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participants (8)
-
Anthony Minessale
-
Brian West
-
Ivan C Myrvold
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Martin Joseph
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Michael Collins
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Michael Jerris
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Rupa Schomaker
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Torstein Knutsen