Re: [Asterisk-Users] IAX2 Max Retries dropped calls Firefly
There's an update to Firefly on Virbiage http://www.virbiage.com/firefly/download/firefly-thirdparty.exe lots of bug fixes - see if that helps -Adam Paul Redstone wrote: Hi We've recently set up and are using with success 1.0.7 using a Junghanns quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works very well, however we're getting cases where sometimes the call just drops. From setting more verbose modes we get a log which is shown below. The problem seems to be the maxretries message which comes from chan_iax2. We are using Firefly 1.9.8 build 3945. However I cannot work out what this message means. There is some suggestion in when it occurs that it might be an IP connection issue from the softphone to the asterisk box. Connection is in one office via 100 M switches, very simple direct path. Firefly running Windows XP SP2. We're planning to try another softphone but quite like Firefly. Can anyone advise on this? Thanks Paul === Log extract -- Hungup 'Zap/1-1' == Spawn extension (bodiam, NN, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/ 10' -- Hungup 'IAX2/[EMAIL PROTECTED]/10' -- Registered '355' (AUTHENTICATED) at -- Registered '354' (AUTHENTICATED) at -- Accepting AUTHENTICATED call from requested format = 1024 , actual format = 1024 -- Executing Macro(IAX2/[EMAIL PROTECTED]/11, bodiam-iaxsip|352|IAX2/352) in new s tack -- Executing Dial(IAX2/[EMAIL PROTECTED]/11, IAX2/352|20|tT) in new stack -- Called 352 -- Call accepted by (format ilbc) -- Format for call is ilbc -- IAX2/352/15 is ringing -- IAX2/352/15 answered IAX2/[EMAIL PROTECTED]/11 -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/11 and IAX2/352/15 May 17 11:47:56 WARNING[2763]: chan_iax2.c:1480 attempt_transmit: Max retries ex ceeded to host on IAX2/[EMAIL PROTECTED]/7 (type = 6, subclass = 2, ts=3800 76, seqno=66) May 17 11:47:56 WARNING[2763]: chan_iax2.c:1480 attempt_transmit: Max retries ex ceeded to host on IAX2/[EMAIL PROTECTED]/7 (type = 6, subclass = 11, ts=380 079, seqno=67) -- Hungup 'Zap/2-1' == Spawn extension (bodiam, NN, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/ 7' -- Hungup 'IAX2/[EMAIL PROTECTED]/7' -- Hungup 'IAX2/352/15' == Spawn extension (macro-bodiam-iaxsip, s, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/11' in macro 'bodiam-iaxsip' == Spawn extension (bodiam, 352, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/11' -- Hungup 'IAX2/[EMAIL PROTECTED]/11' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 codec
Ivan Meic (Vox Mundi) wrote: Actually G.729A is a reduced complexity version, and G.729B is a version with silence suppression. The data rate while sending voice is exactly the same, although the quality of G.729B should be a little higher. However the average rate for B can be lower if the silence suppression is used. Right now Asterisk doesn't make use of that silence suppression, so it makes not difference. Steve, Any Cisco gateway support two G.729 variants. They call them g729r8 and g729br8. So I guess that Cisco never implemented a reduced complexity version ? Also as far as I understand there are 3 G.729 variants generaly used. The first version (G.729), Annex A and Annex B. Are they all compatible with each others ? Ivan Actually, I believe Cisco uses G.729A, as they use TI chips. The difference between G.729 and G.729A is Annex A spends less time looking for the optimal representation of the voice. G.729 and G.729A are completely compatible. Annex B adds silence suppression. I believe you need to support and negotiate annex B on both ends if you want to use it. I'm against silence suppression but that's just me. Annex C is the floating point version and obviously completely compatiable as well -Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Quality
What's your end device? if it's a voip device (eg SIP phone or a soft phone) then you shouldn't need a jitter buffer. Also, you don't need bandwidth=low if you specify the codecs (the disallow=all will override the bandwidth=low) and maxjitterbuffer is the param you're after with this line jitterbuffer=200 I'm guessing -Adam [EMAIL PROTECTED] wrote: Hello, I have setup two * servers and they are communicating using IAX. I'm passing calls from SRV A (internet connection T1) to SRV B (internet connection: 512). For some reasons I have an issue with the quality. The voice is a bit scratchy. I have tried iLBC and SPEEX, but it didn't make any difference. Now, assuming that I have an issue with Bandwidth, what would be the best way to configure my iax.conf. (A bit confused about jitterbuffer and tos) Here is my iax.conf @ location A: [general] port=4569 bandwidth=low disallow=all allow=ilbc ;allow=ulaw ;allow=speex jitterbuffer=200 jitterbuffer=yes tos=lowdelay and iax.conf @ location B: [general] port=4569 bandwidth=low disallow=all allow=ilbc ;allow=ulaw ;allow=speex jitterbuffer=200 jitterbuffer=yes tos=lowdelay [guest] type=user context=default callerid=Guest IAX User disallow=all allow=ilbc Thanks guys ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX aproprietary protocol
Steve Kann wrote: Something *proprietary* is something exclusively owned by someone nobody owns the IAX2 protocol. Although, Digium have trademarked IAX ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Third party Firefly issue very weird??
Ethereal on various boxes should help solve the issue (probably firewall) Jon Walsh wrote: When I connect to the third party softphone (firefly) I get connected at my house and at my office where I have the asterisk..but when I went to my friends house to set him up his firefly showed a gray circle like it was not connecting at all? Has Anyone seen this happen what is causing this no to connect, does anyone know ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SIP vs. IAX
you mean IAX isn't a standard :) Also IAX requires your call router / billing gateway to handle the voice traffic too (or you put your CDR recording at the end points) With SIP, just the signaling is needed, allowing more scalability. I recall talking about this at astericon but it never eventuated to anything. The idea was basically keeping the original channel open even on a native transfer. -Adam Tom Samplonius wrote: On Mon, 14 Mar 2005 16:47:21 -0700, Joseph [EMAIL PROTECTED] wrote: * SIP isn't a standard. It could be made into an official standard, if there was a standards document. Someone should write one, and start an IETF working-group. If the IETF adopted it, there would be wider acceptance. * SIP NAT traversal in Asterisk is harder than it needs to be. This should be getting better. But SIP in general isn't very easy to configure in Asterisk. It sounds like this is getting a lot better in the next release (no more goofy peer vs. friend distinction). Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
Duane wrote: Adam Hart wrote: As always, I'm happy to announce a new version of Firefly. Firefly 1.9.8 has more of what you want and less of what you don't http://www.virbiage.com/firefly/download/firefly-thirdparty.exe There's a few bug fixes - notably fixed the Reject button and sending of audio before answering in some circumstances. Has anyone been able to make firefly work under wine at all? If so how? A decent linux client is the only thing skype has over SIP/IAX... Few people have claimed success, I'm not sure how though. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Firefly version
As always, I'm happy to announce a new version of Firefly. Firefly 1.9.8 has more of what you want and less of what you don't http://www.virbiage.com/firefly/download/firefly-thirdparty.exe There's a few bug fixes - notably fixed the Reject button and sending of audio before answering in some circumstances. -Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip to IAX ok, ZAP to IAX FAILS
use ethereal or iax2 debug to see what capabilities are been set in your NEW message Ernie Ankele wrote: Hello, Could someone give me clues where to figure out this problem? If I call from a Sip client to an Firefly client running IAX, the call connects fine, no problems. I can connect to asterisk using any codec (excepting g.729) on firefly to voicemail and music-on-hold, other sip extensions and everything works fine. If I try to connect to the same client via a ZAP channel (X100P clone), via Dial(IAX2/) I get an error : Jan 10 18:07:26 WARNING[1378]: chan_iax2.c:6017 socket_read: Call rejected by xx.xxx.xxx.xxx: No compatible Codecs I just updated asterisk to cvs-head-1-10-2005. All codecs are allowed in IAX.conf and all codecs are enabled on Firefly. I have tried everything I can think of- only enable gsm, only gsm+G.711, all codecs on firefly. Same results. Anyone else with this issue? Thanks, Ernie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip to IAX ok, ZAP to IAX FAILS
Can you paste the full NEW frame please. Could be Preference vs capability thanks, Adam Ernie Ankele wrote: On a sip to iax : CODEC_PREFS : (gsm|ulaw|alaw|ilbc) and Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 0ms SCall: 19170 DCall: 1 [xx.xxx.xxx.xxx:20406] FORMAT : 4 -- Call accepted by xx.xxx.xxx.xxx (format ulaw) -- Format for call is ulaw On ZAP to IAX: CODEC_PREFS : (gsm|ulaw|alaw|ilbc) and Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 0ms SCall: 15725 DCall: 3 [xx.xxx.xxx.xxx:20406] CAUSE : No compatible Codecs Jan 10 18:50:06 WARNING[1378]: chan_iax2.c:6017 socket_read: Call rejected by xx.xxx.xxx.xxx: No compatible Codecs Thanks, Ernie On Jan 10, 2005, at 6:34 PM, Adam Hart wrote: use ethereal or iax2 debug to see what capabilities are been set in your NEW message Ernie Ankele wrote: Hello, Could someone give me clues where to figure out this problem? If I call from a Sip client to an Firefly client running IAX, the call connects fine, no problems. I can connect to asterisk using any codec (excepting g.729) on firefly to voicemail and music-on-hold, other sip extensions and everything works fine. If I try to connect to the same client via a ZAP channel (X100P clone), via Dial(IAX2/) I get an error : Jan 10 18:07:26 WARNING[1378]: chan_iax2.c:6017 socket_read: Call rejected by xx.xxx.xxx.xxx: No compatible Codecs I just updated asterisk to cvs-head-1-10-2005. All codecs are allowed in IAX.conf and all codecs are enabled on Firefly. I have tried everything I can think of- only enable gsm, only gsm+G.711, all codecs on firefly. Same results. Anyone else with this issue? Thanks, Ernie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How expensive are the different codecs? (Regarding CPU time)
Michael Vogel wrote: Hi! The encoding, decoding and recoding cost cpu time, that's sure. But does this time differs much depending on the used codec? Is - for example - a G729 faster than a GSM codec? Try 'show translations' in asterisk's CLI (GSM is much faster than G.729) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] drive space for voice mail
Christopher L. Wade wrote: Matthew Boehm wrote: Can you say 'overkill' ? *smiles* I just recorded a 2min voicemail and the resulting file on the server was slightly over 200KB in size. We are only storing 1 format of soundfiles, WAV49. A 160GB drive is approx 1,677,721,160 KB. At the rate above you would be able to store almost 28,000 hours of voicemail messages. Someone wanna check my math? Unless my recent math [280,000 hours] was wrong, thats ~ 31 years of voicemail :) Better get 200GB just to be safe :p ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no plain text passwords in iax.conf
Bastian Schern wrote: Hello Asterisk friends, is it possible to avoid plain text passwords in the iax.conf or the iaxfriends MySQL database table? Asterisk needs the plain text password to authenicate. You could wrap a base64 decode when reading the passwords, but this is obsecurity, yet simple to implement should prevent the casual browser. I guess a more secure method would public key crypto and give asterisk the key at runtime (obviously not 100% secure either) -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no plain text passwords in iax.conf
Bastian Schern wrote: Adam Hart schrieb: Bastian Schern wrote: Hello Asterisk friends, is it possible to avoid plain text passwords in the iax.conf or the iaxfriends MySQL database table? Asterisk needs the plain text password to authenicate. You could wrap a base64 decode when reading the passwords, but this is obsecurity, yet simple to implement should prevent the casual browser. I guess a more secure method would public key crypto and give asterisk the key at runtime (obviously not 100% secure either) I found out that MySQL offers some methods to store strong passwords: http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers But how I use this with Asterisk? That's using private key crypto, when you store the password you do aes_encode(password,somekey) then when asterisk reads it, do a aes_decode(password,somekey) - modify chan_iax2 when you do the select - change the SQL statement: the column 'secret' to 'aes_decode(secret,somekey) as real_secret' then below change secret to real_secret. good luck, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no plain text passwords in iax.conf
Bastian Schern wrote: Adam Hart schrieb: Bastian Schern wrote: Adam Hart schrieb: Bastian Schern wrote: Hello Asterisk friends, is it possible to avoid plain text passwords in the iax.conf or the iaxfriends MySQL database table? Asterisk needs the plain text password to authenicate. You could wrap a base64 decode when reading the passwords, but this is obsecurity, yet simple to implement should prevent the casual browser. I guess a more secure method would public key crypto and give asterisk the key at runtime (obviously not 100% secure either) I found out that MySQL offers some methods to store strong passwords: http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers But how I use this with Asterisk? That's using private key crypto, when you store the password you do aes_encode(password,somekey) then when asterisk reads it, do a aes_decode(password,somekey) - modify chan_iax2 when you do the select - change the SQL statement: the column 'secret' to 'aes_decode(secret,somekey) as real_secret' then below change secret to real_secret. What is about the field md5secret similar to sip.conf? Is that not a solution for iax.conf? (To the best of my knowledge) sip does md5 authenication differently and doesn't require the actual password, just the md5 of it (and user domain). Iax requires it to md5 with the challenge. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gentoo and Asterisk - any experiences?
Niels Chr. Sørensen wrote: Hi, In constant search for optimization, a friend told us about his experience with Gentoo Linux-distro. He claimed that he doubled the performance of his server by changing to Gentoo from Debian. Does anyone have any experience with running Asterisk on a Gentoo linux? Your friend is a Gentoo hippie - a lot of people use Gentoo with Asterisk (myself included) but you won't see a noticable performance difference with Asterisk, if any at all. (Lets not start a flame war) -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP to IP call without server?
nkb wrote: Hi. I'm really new. I was just wondering if it is possible at all to do a IP to IP call without a * server (or as a matter of fact, any other kind of server)? say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at hisdomain.com's 192.168.0.3. Is this sort of things possible? Or must we all both be registered with the same server to do that? Can this not be done without passing thru server (*)? Thanks. Firefly, along with most softclients, you can do this - dial iax2/ip or sip/ip - F2 is a good shortcut for dial URL ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly on Linux
Andrew Kohlsmith wrote: On November 23, 2004 05:28 pm, Adam Hart wrote: iptables -t nat -I PREROUTING -p udp -d EXTIP --dport 4569 -m state --state NEW -j DNAT --to-destination ASTIP iptables -t nat -I POSTROUTING -p udp -d ASTIP --dport 4569 -j MASQUERADE Any reason why you need both these statements instead of just a single iptables -t nat -I PREROUTING -p udp -d EXTIP --dport 4569 -j DNAT --to-destination ASTIP oops, no need - I was thinking one interface, so the packets would come back through it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly:Canreinvite problem
Run ethereal and look the dump, prehaps A) the SIP invite doesn't match the correct IP port B)try turning on Asterisk's NAT fix C) send the dump to me :) -Adam Alejandro Gutiérrez wrote: Hi!. I am testing firefly and I can say it's a great program, but I have a problem. When I use Sip and I activate the canreinvite option in Asterisk, I can't hear anything. My network is the following: -Two Firefly clients with SIP. Each firefly is in different networks behind NAT. -One Asterisk server with a public IP. First, I tested my network with canreinvite=no. Everything was perfect, the voice quality was quite good. After that, I changed to canreinvite=yes, and I could't hear anything. I thought that my routers might be stopping the voice streams, but I ran Ethereal and I could see the voice was arriving to my boxes. With IAX, canreinvite works but nowadays SIP phones are majority :(. Any ideas?. Thanks in advance. __ Renovamos el Correo Yahoo!: ¡100 MB GRATIS! Nuevos servicios, más seguridad http://correo.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly on Linux
An untested guess iptables -t nat -I PREROUTING -p udp -d EXTIP --dport 4569 -m state --state NEW -j DNAT --to-destination ASTIP iptables -t nat -I POSTROUTING -p udp -d ASTIP --dport 4569 -j MASQUERADE cheers, Adam Peter Osborne wrote: Hello, With all the talk about Firefly, I decided to check it out, it seems to work under wine (IAX only for some reason) so I'm thinking about using it on the road. Now, my Asterisk box is behind a firewall, so I have set the firewall to forward UDP port 4569 to my Asterisk box put I'm having problems with this. I followed the instructions on the Asterisk Firewall Rules page but it seems to a slightly different setup I guess. Does anyone have an iptables setup that will accept and forward IAX2 traffic from an external box to a box on the private network? Thanks, Pete ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Firefly Problems
Chris Olson wrote: Chris Olson wrote: Hello, I have firefly installed and it is somewhat working. It is registering with my Asterisk server and I can call out, but I receive no audio coming into Firefly. From the Asterisk end, everything looks OK with the call, just no audio is being received on the Firefly end. I am using 1.9.6 Any ideas? a fix for this will be out tommorrow - you can temporarily fix it by inserting the r option into your dial cmd cheers, Adam Thanks Adam. Can you let us know when the fix is available and where we can download the fixed 3rd-party from? A little more info ... this is actually a one-way audio problem as audio passes from Firefly to Asterisk, but not from Asterisk to Firefly. You can grab the new one from http://www.virbiage.com/ now, if anyone wants the old version, it's at http://www.virbiage.com/firefly/firefly-thirdparty196.exe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Firefly Problems
Chris Olson wrote: Thanks Adam. Can you let us know when the fix is available and where we can download the fixed 3rd-party from? A little more info ... this is actually a one-way audio problem as audio passes from Firefly to Asterisk, but not from Asterisk to Firefly. You can grab the new one from http://www.virbiage.com/ now, if anyone wants the old version, it's at http://www.virbiage.com/firefly/firefly-thirdparty196.exe Thanks Adam. The new version is working very well. I really like this softphone. Since you are doing such a great job on this :), I have 1 more question. (1) Is there a way to pre-configure the phone for some clients? By this I mean by having some extensions pre-programmed and also having the IAX option pre-programmed? I have some people I'd like to do beta testing with and was hoping I could just send them the installation file, and it would be preconfigured for them and ready to go. Thanks, Chris IAX settings are in the registry, contacts are in contacts.txt in the program's directory... which makes it hard, hey? :) You could wrap a Nullsoft installer around it or give a .reg contacts.txt file I think rendenvous support would help here, well in the provisioning sense anyway. It's coming soon. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly problems
Chris Olson wrote: Hello, I have firefly installed and it is somewhat working. It is registering with my Asterisk server and I can call out, but I receive no audio coming into Firefly. From the Asterisk end, everything looks OK with the call, just no audio is being received on the Firefly end. I am using 1.9.6 Any ideas? a fix for this will be out tommorrow - you can temporarily fix it by inserting the r option into your dial cmd cheers, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly 1.9.5 and 20041117 CVS HEAD -- IAX2 one way audio
try running ethereal, make sure everything looks ok and send me the result. No firewall? Also, download debugview from www.sysinternals.com to see Firefly's debug msgs. Could be simply wrong audio device? Andrew Kohlsmith wrote: Using Firefly 1.9.5 (thirdparty) on Win2k Using Asterisk CVS HEAD 20041117 (also tried with 20040806 and 200410-something) IAX2, no NAT. Firefly-Asterisk audio works, but I can't hear anything from the other side. Using GSM codec, also tried ulaw. Any ideas? -A. relevant bits of iax.conf: [andrew-bt] type=peer host=dynamic trunk=no [andrew-bt] type=user context=fxs secret=12345 host=dynamic trunk=no ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UDP Fragmentation Problem
Andrew Kohlsmith wrote: On October 31, 2004 05:36 pm, Bastian Schern wrote: I've got no success to get a friend in Bogota (Colombia) connected to my Asterisk. He has got a ISDN Internet connection and the UDP packets will be fragmented. It seems that the MTU of this connection is round about 400 to 500 Bytes. Therefore most UDP-SIP packages are fragmented. Is Asterisk not able to handle fragmented UDP packages? Is it possible to use SIP over TCP with X-Lite? Or has somebody another hint for me? As far as I am aware there is no such thing as a fragmented UDP packet; each packet is sent out on its own, there is no coherency between UDP packets like there is with TCP packets. I could be very wrong here, it's been a late night with the kids. :-) Packet fragmentation is at the IP layer, so UDP will have fragmented packets too. But... the OS should handle that and Asterisk shouldn't find out - it's a all or none policy, so it should receive the whole packet at once or nothing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can bad person with SIPp attack Asterisk ?
[EMAIL PROTECTED] wrote: Hello I would say, First of all, for users who are authenticated, so really can make calls, just configure asterisk to limit the number of calls users can make concurrently Next, put a firewall in front of your asterisk box which rate limits the number of connection attempts per second per host.. If you limit this to lets say about 25 to 50 connection attempts per second per host I would say you're pretty safe and your asterisk box can't really get overloaded with malicious packets. this burst limit depends on your config as you might get much traffic from certain IP's ofcourse Niels With SIP and IAX, it's UDP (* doesn't do TCP SIP) you can spoof the source address. An attack similar to TCP SYN attack would work. Actually there's better attacks I can think of. Low cpu auth replys would partly solve it with IAX, moving to TCP (even TLS) with SIP is much safer. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GPL thoughts
Remember the requirements of GPL is regarding distribution, not use, you can do what ever you like with it internally, with no requirement to publish it. Config files being GPL doesn't really make sense as you would only ever be distributing them as they are anyway (not compiling them) GPL in a simple sense (feeling free to correct me on this) is if you give some the binary of software containing source under GPL, you must also give them the source -Adam Ronald Wiplinger wrote: I have just a quick question: Are the configuration files are covered from GPL ??? I doubt so, but would like to make sure. The configuration files (/etc/asterisk/* ) include passwords, which I hardly would like that it must be public ;-) My thinking is to get my work somehow paid, by creating special configuration files for special solutions. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GPL thoughts
Ronald Wiplinger wrote: On Tuesday 26 October 2004 12:33, Adam Hart wrote: Remember the requirements of GPL is regarding distribution, not use, you can do what ever you like with it internally, with no requirement to publish it. Config files being GPL doesn't really make sense as you would only ever be distributing them as they are anyway (not compiling them) GPL in a simple sense (feeling free to correct me on this) is if you give some the binary of software containing source under GPL, you must also give them the source Thanks Adam! Now lets think one step further. If we add a patch for the program to read some of the configuration files, which are encrypted. This patch would be brought back to the open source community and if they accept it, it could be implemented. If they don't you still can get the patch from other places. The patch opens with a key the encrypted file and checks against the registration server if the script is licensed to the customer. If yes, everthing is ok, if not than the system can still use the script for demo purposes for one hour. What is with that thought? Note that I changed the word from configuration file to script, which could be an external program, called by the configuration file. May I suggest the key remains on the registration server and the registration server returns the key if they are licensed, otherwise people could easily cut out the reg server. Although, they couldn't have the script for demo purposes. A closed source daemon (like macrovision license manager) is your next safest bet. I'd suggest completely revisiting the solution. Service contract, leasing, etc regarding config and gpl, I would think your app would be generating config's from scratch anyway? Besides that, are you asking that if only having the encrypted copy of the config on their computer against GPL? I'd say no, but i'm no laywer (GPL is designed for source code) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FireFly w/ SIP
The best way for me or yourself to debug it is using ethereal (google for it) and debugview from www.sysinternals.com. I'm happy to help, so send the logs, the native transfer might be the issue. -Adam Willem de Groot wrote: Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk? It works in IAX mode, but in SIP mode I am unable to hear anything (no dialtone, no voice). I am able to setup a conversation with another SIP phone though (Xlite, Grandstream) and the other side can hear the FireFly user just fine (both sides using g711u). I tried different PC's with different audio hardware. They all work fine using FireFly in IAX mode and using other softphones, so I guess it must be related so FireFly in SIP mode. This is my SIP config: [201] type=friend host=dynamic dtmfmode=rfc2833 context=sip canreinvite=yes FireFly is also configured for rfc2833 dtmf. Thanks for any suggestions! Willem ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FireFly SIP Registration Interval
We'll add that to next version, should be out next week Deon Rodden wrote: I put FireFly on my moms computer, but ran into a problem. She went home and was able to place calls from it (using her headset and such). But, she could not receive calls. I figured out the problem was with the registration, firefly doesnt re-register often enough, so the connection gets stale and the NAT Device forgets about the connection, so no new incoming calls can be made. I put X-Lite on her computer and changed the re-registration interval from the default of 3600 to 60 seconds. Now I can call her anytime. But, theres choppiness on the line. Her ability to transmit/upload/send voice to me is bad, I hear choppiness and such. FireFly worked fine, no choppiness, same router, same connection. I tried X-Lite and FireFly on my laptop but both perform equally. I like the simplicity and interface of firefly, its nicer, anybody know of a way to change the sip registration interval? Anybody know of another program other than x-lite or firefly? One that doesnt have problems sending audio and one that allows you to change the sip registration interval? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone - PBX Phone
Seems to be alot of these questions on the mailing list recently. AUSTEL is the old name for the ACA, A-tick is the correct term for certification. It's only illegal if you connect to a carrier network without A-tick (you can get consent from them to connect without A-tick). The ACA has plently of info, such as http://www.aca.gov.au/consumer_info/fact_sheets/consumer_fact_sheets/fsc69.htm -Adam P J wrote: Thanks Paul. I've been getting conflicting information about Austel permits.. Can any one confirm that the card connecting Asterix to an existing PABX does not require Austel approval? Therefore, I could use a simple 1 port Compatible X100P FXO Card (that doesn't have Austel approval)? Thanks. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Friday, September 17, 2004 3:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP Phone - PBX Phone The TDM400 is used for both PSTN and PABX - PABX connections, from memory. The card only requires an Austel permit if it is to be connected to an outside line, from memory. Cost wise, you can get the TDM400 with 1 line for less than $200, or about $500 with 4 lines hooked up to it. Later, Paul Hales IT Support Adairs -Original Message- From: Phil Stevens [mailto:[EMAIL PROTECTED] Sent: Friday, 17 September 2004 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP Phone - PBX Phone Hi Paul, I have yet to find out the make and model of the PABX, I was just doing some general background research at this point. Is the same device (TDM400?) used for connecting both to PSTN and to other PABX's? I don't need to connect the Asterisk box to PSTN at this point, just to another PABX. I do know that the current PABX already has a tie-line that connects to a second PABX. If Asterisk connects to another PABX via FXO ports, does the card have to have an Austel permit? I do not want to connect Asterisk to PSTN, only to another PABX. Do I have to purchase an $800 4-port FXO Austel-certified card? Or, is there a cheaper option (1 port? for testing purposes) that will satisfy Austel requirements? Thanks. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Friday, September 17, 2004 2:02 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP Phone - PBX Phone Good to see another Australian user on the list! You could set up a card with some FXO ports (TDM400?) and use those lines to hook up the Asterisk box to your existing PABX. But I am sure someone else will come up with a _much_ more clever solution. Later, PaulH Melbourne -Original Message- From: P J [mailto:[EMAIL PROTECTED] Sent: Friday, 17 September 2004 12:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP Phone - PBX Phone Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call each other, however - - I would also like to link Asterisk to our existing PBX so that SIP phones could call standard phones on our existing PBX system (and vice-versa). - I *do not* need to use Asterisk to call out via PSTN or ITSP. All outbound calls will be via the existing PBX. What hardware device is required to link the Asterisk box to the existing PBX? Could the SIP phones call the standard phones on our existing PBX system? If so, how does Asterisk do this? Thanks in advance. Ps. Even though I *do not* need to use Asterisk to call out via PSTN, what hardware device would be required to do this? And, how does this device differ from the device that links Asterisk to the existing PBX? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] forwarding calls thru Freshtel
register on gateway.freshtel.net, not cts-au -for firefly numbers, call gateway.freshtel.net -for PSTN termination, call cts-au.freshtel.net don't think you need the @freshtel at the end of your dial good luck, Adam Shaun Dwyer wrote: Hi, I'm having some problems getting calls to go out via freshtel. There dosn't seem to be any specific information on how to get it working anywhere. The only information I've found is here: http://www.voip-info.org/wiki-Freshtel and that dosn't give you any idea of how to actually get it working. I've tried adapting information from other IAX2 provider examples but have yet to find a working solution. in my iax.conf I have: register = freshtel_number:[EMAIL PROTECTED] [freshtel] type=friend host=cts-au.freshtel.net secret=password context=from-freshtel qualify=yes In my extensions.conf, I have: exten = _99.,1,StripMSD,2 exten = _99.,2,(Dial(IAX2/freshtel_number:[EMAIL PROTECTED]/[EMAIL PROTECTED]) The general idea is to dial 99 followed by the number to dial thru freshtel. In my SIP client phone (X-Lite) I get 'call failed: 403 Forbidden' on the asterisk server console I get: -- Executing StripMSD(SIP/101-b0d9, 2) in new stack Sep 7 14:42:55 WARNING[110756]: pbx.c:1872 ast_pbx_run: Channel 'SIP/101-b0d9' sent into invalid extension '88669100' in context 'intern', but no invalid handler I have multiple SIP phones, all can dial eachother as well as the echo test extension I've setup. They can also interact with voicemail with no problems. I also have a X101P card setup connected to a PSTN line and I can make calls though that OK as well. There is an echo problem with the X101P, but thats another story... Any one have any suggestions with regards to my freshtel problem? I've yet to try a SIP connection to them. I'll be trying that later thisarvo. Cheers, -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disconnection From IAXTel
Dave Cotton wrote: On Sat, 2004-08-28 at 23:36 -0700, Muiz Motani wrote: I'm using the firefly third-party softphone. However, the same thing happened when I used IAXphone 2.0. I can't offer any real solution because I was only testing the connection with Firefly, but I got exactly the same symptoms whenever the Firefly softphones tried to communicate, with or without the notransfer= setting. I blamed it on the Firefly but you say the same thing happens with IAXphone. Once hardphones where in place that problem went away. I believe this was discussed awhile ago, the solution was to set qualify=no - if anyone knows why it's happening, I'm happily fix it in firefly -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with mysql and with asterisk
try installing mysql-devel -Adam DIPAK PAUL wrote: Hi Every one and Lerale Erwan I have briefly describe my problem and I have provide the steps as follows: I have intalled redhat properly and from the konsole I checked with mysql. rpm -qa | grep mysql and the konsole provide me the message: mysql-3.23.54a-11 mysql-server-3.23.54a-11 Then I have download the asterisk and addons: By the using of : cd /usr/src export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login - the password is anoncvs. cvs checkout asterisk Then cd/usr/src export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login - the password is anoncvs. cvs checkout asterisk-addons Compile /usr/src/asterisk-addons as follows: cd asterisk-addons make clean make install But the system send me an error messge like ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory for x in ; do install -m 755 $x /usr/lib/asterisk/modules ; done The I have used to use mysql-vm-routines, set USE_MYSQL_VM_INTERFACE to 1 in asterisk/apps/Makefile , then put this file into asterisk/apps/ and (re)build asterisk. Then use make from the /usr/src/asterisk/ Then system have give me this type of error message: make[1]: Entering directory `/usr/src/asterisk/apps' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\CVS-HEAD-08/23/04-11:39:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -DUSEMYSQLVM -c -o app_voicemail.o app_voicemail.c app_voicemail.c:45:25: mysql/mysql.h: No such file or directory In file included from app_voicemail.c:372: mysql-vm-routines.h:7: parse error before '*' token mysql-vm-routines.h:7: warning: type defaults to `int' in declaration of `dbhandler' mysql-vm-routines.h:7: warning: data definition has no type or storage class mysql-vm-routines.h: In function `mysql_login': mysql-vm-routines.h:18: warning: implicit declaration of function `mysql_init' mysql-vm-routines.h:18: warning: assignment makes pointer from integer without a cast mysql-vm-routines.h:19: warning: implicit declaration of function `mysql_real_connect' mysql-vm-routines.h: In function `mysql_logout': mysql-vm-routines.h:29: warning: implicit declaration of function `mysql_close' mysql-vm-routines.h: In function `find_user': mysql-vm-routines.h:35: `MYSQL_RES' undeclared (first use in this function) mysql-vm-routines.h:35: (Each undeclared identifier is reported only once mysql-vm-routines.h:35: for each function it appears in.) mysql-vm-routines.h:35: `result' undeclared (first use in this function) mysql-vm-routines.h:36: `MYSQL_ROW' undeclared (first use in this function) mysql-vm-routines.h:36: parse error before rowval mysql-vm-routines.h:37: `MYSQL_FIELD' undeclared (first use in this function) mysql-vm-routines.h:37: `fields' undeclared (first use in this function) mysql-vm-routines.h:68: warning: implicit declaration of function `mysql_query' mysql-vm-routines.h:69: warning: implicit declaration of function `mysql_store_result' mysql-vm-routines.h:70: `rowval' undeclared (first use in this function) mysql-vm-routines.h:70: warning: implicit declaration of function `mysql_fetch_row' mysql-vm-routines.h:71: warning: implicit declaration of function `mysql_num_fields' mysql-vm-routines.h:72: warning: implicit declaration of function `mysql_fetch_fields' mysql-vm-routines.h:89: warning: implicit declaration of function `mysql_free_result' make[1]: *** [app_voicemail.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Please help me. I had totaly upset to install the asterisk with cdr. Please help me because i am now helpless. With best regards. Dipak Kumar Paul Tryarc LLC _ Claim your Citibank Ready Cash today. http://go.msnserver.com/IN/54177.asp Its fast, easy and affordable. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec
Daniel Niasoff wrote: Hi Everyone, Is G729 more sensitive to packet loss or delays due to its higher compression. If Ive generally got the bandwidth available, am I best sticking to ulaw. G.729 has lost packet concealment, G.711 doesn't. G.711 will sound better otherwise if you can afford the bandwidth. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec
Steve Underwood wrote: Adam Hart wrote: Daniel Niasoff wrote: Is G729 more sensitive to packet loss or delays due to its higher compression. If Ive generally got the bandwidth available, am I best sticking to ulaw. G.729 has lost packet concealment, G.711 doesn't. G.711 will sound better otherwise if you can afford the bandwidth. Eh? G.729 has no particular features to allow more effective packet loss concealment. iLBC has, but at the cost of a substantially higher bit rate. In fact G.711 is a little ahead of G.729 in the regard, since packets are completely independant. The smoothing in G.729 means you need the previous packet to decode the current one properly. Regards, Steve I believe you're mistaken - G.729 works similar to iLBC and speex. iLBC works better as the packets are independent but G.729 still has a function for packet loss concealment. prehaps have a look at http://www.speex.org/comparison.html -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium G729 codecs
just send me your key and I'll help :p just kidding try ftp://ftp.digium.com/pub/asterisk/g729/ the README and the needed files are in there Jean-Yves Avenard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dear all Last week I purchased 10 G729 codec licenses from Digium. The only thing I got from them was the invoice. No license file nothing. After chasing them up, I got an other email giving me something that looks like this: asteriskpbx-600x:G729-xx they told me a README attachment file was provided but there was nothing. Is that all I need? i looked on google, wiki to no available.. If yes, how can I install this? Regards Jean-Yves -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBBaEAXeDVKqIr3GURAsIMAJwOKKvFmGwiwWFshECracsBYTIS3QCfSMQe 7MxmGKgCyIJlNYXQ5jXsT5U= =G5mX -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 codec doesn't seem to work *even* after installing the license
try sip debug and see what each side is offering in codecs (make sure yo u have allow=g729 Walter Klomp wrote: Hi, I am trying to post this again as I am getting no answers and the [EMAIL PROTECTED] bounces... (I have searched the whole list and can't find the answer either) I have installed a 5 user license for G.729 and want to route calls through Asterisk from my G.729 phone to Cisco AS5300 also using G729. Both Cisco and the phone connect using this codec if I do not force the call to go through * However if I say canreinvite=no in the sip.conf for either of these gadgets, the call will fail with No compatible codecs! I have bought a 5 user license just to try and fix this, apparently it doesn't work. I want to protect the Cisco gateway from unauthorized use, but still using a cost-effective codec such as g.723 or g.729 ? [codec_g729a.so] = (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 5f:a1:18:82:47:6f:a8:f7:33:4e:7d:77:e8:1d:60:15:53:ec:49:aa == Found license 'G729-700241AB' providing 5 channels == Found total of 5 G.729 licenses == Registered translator 'g729tolin' from format G729A to SLINR, cost 2 == Registered translator 'lintog729' from format SLINR to G729A, cost 12 I was hoping by letting it ring out, I would get a voice-mail message, but that doesn't work either... *CLI Jul 13 11:29:08 WARNING[98310]: chan_sip.c:2696 process_sdp: No compatible codecs! -- Executing Dial(SIP/67.23.212.25-0814f830, SIP/334|20) in new stack -- Called 334 -- SIP/334-26f8 is ringing -- Nobody picked up in 2 ms -- Executing VoiceMail(SIP/67.23.212.25-0814f830, u334) in new stack -- Playing 'vm-theperson' (language 'en') == Spawn extension (default, 4084, 2) exited non-zero on 'SIP/67.23.212.25-0814f830' I have dropped this question at the asterisk user list some days ago, but it's being ignored... (or nobody has the answer) Can anybody shed some light on this ? Warmest Regards, Walter Klomp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crashing with no indication why.
Daniel Daley wrote: I'm hoping someone might have seen this before because I'm just about at a loss of what to do. I have an asterisk system setup in a call center environment with multiple queues. After a random uptime asterisk will suddenly come to a partial halt where I can connect to the cli but issuing a command such as show channels gives no response, and calls cannot be made in or out. Calls in progress usually drop as well, but if they don't right away, after a minute or so they will. To remedy the problem I have to do a restart on asterisk, which of course makes all the agents have to login again and is just a big mess. I have agents being dynamically added to the queues via an AGI script, also the agents are added to all queues so that they can take calls from any of them. I'm not sure if this is important but since I use the AgentCallbackLogin function I have all the agents inside their own context so that I can use a macro to determine if they are on an outgoing call (using app_checkgroup) before ringing them to prevent call waiting tones. I've thoroughly searched the messages log, in which I have both verbose and debug logging enabled. I've never found anything to indicate a problem, it simply looks like calls just slow down and stop. One other thing that may be important, I have a daemon running which stays connected to the manager api listening for events and sending off two commands every 10 seconds, one to get the status of the queues, and one to get the status of agents. My cvs version is CVS-HEAD-06/24/04-06:49:37. I've looked through all the latest cvs updates and bug reports and don't see anything that would be related. Has anyone seen this before, can anyone suggest anything I might try? With both being unable to reproduce this at will and the lack of messages or log entries pointing to the problem I'm pretty much up against a wall. Thank you for any help anyone can offer, --Daniel Daley-- [EMAIL PROTECTED] Depending on your dev skills, you could run asterisk in gdb and then look at the status of each thread when the problem occurs. Other than that, try an older version of asterisk PS Please don't post in both lists, it isn't a dev question -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP hackers gut Caller ID
Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. http://www.theregister.co.uk/2004/07/07/hackers_gut_voip/ I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Randy Bush is a destructive force with a hidden professional agenda
Bradley D. Thornton wrote: snip i don't need nats, nat traversal, nat anything. if i did, iax might well be one of the technologies i would consider. but i don't. snip Watch out for this man Bush! He is a professional espionage troll and hides his agent status behind his condescending facade. What do you want Randy? Go back to Admiralty way with all your ICANNite cronies if you're not going to behave here! I have no idea who Randy Bush is but I found it funny the first article I found on him was a presentation on why NAT is evil espically for voice. Now he asserts that NAT traversal is not needed. http://www.apnic.net/meetings/17/docs/sigs/policy/addrpol-pres-randy-nats.pdf Can anyone give me a quick rundown on why there's such discontent for this person -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Randy Bush is a destructive force with a hidden professional agenda
[EMAIL PROTECTED] wrote: I have no idea who Randy Bush is but I found it funny the first article I found on him was a presentation on why NAT is evil espically for voice. Now he asserts that NAT traversal is not needed. http://www.apnic.net/meetings/17/docs/sigs/policy/addrpol-pres-randy-nats.pdf I read his message as saying that NAT isn't an problem /he will be facing in what he is aiming to setup/. Given the context around his stuff, I wouldn't be surprised if thats what he meant. / -- emphasis|deemphasis things. Very true, my apologies, I only saw what people had selected in their replies. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Special Delivery from China
Jay Milk wrote: I'm guessing it's too expensive -- looks like my friends took a reference design and barely modified the sample firmware. I was surprised to even find g729 in there (licensing cost), but I'll take it. I'd be glad to get CID name and MWI working, and wouldn't even mind if they dropped H323 and MGCP. I doubt they licensed g.729 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Firefly release - 1.9.3
There's a new firefly release out for those who are using firefly with your lovely asterisk / SIP server. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe the main changes are improved GUI fixes (mouse wheel works now :) ), few url parsing fixes, mic volume control and improved compatibility with SIP servers (namely SER). Send me all bugs, problems and suggestions (even praise if you're feeling generous) -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: I never get to hear more than 5s of the demo channels
I may be wrong but prehaps the answer is in your email -- Executing DigitTimeout(SIP/avenardj-acfc, 5) in new stack -- Set Digit Timeout to 5 -Adam Jean-Yves Avenard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dear all. I'm new to this so please forgive my ignorance if I missed something obvious. I've set-up asterisk 0.9_1 on a FreeBSD 4.10 server (I know it's not linux but that's all we have available at that stage). After some struggle to understand how everything works, I set up some SIP accounts for test purposes. I can log in, make calls to some of the demo system (1234, 1000 etc...) but the playback will always stop after 5s. I mean: I *hear* something (a lady) and after 5 s it stops, and X-lite displays: hung-up On Asterisk console I get the following messages: *CLI Jun 28 08:41:42 NOTICE[135336960]: chan_sip.c:4933 handle_response: Peer 'avenardj' is now REACHABLE! -- Executing Goto(SIP/avenardj-acfc, default|s|1) in new stack -- Goto (default,s,1) -- Executing Wait(SIP/avenardj-acfc, 1) in new stack -- Executing Answer(SIP/avenardj-acfc, ) in new stack -- Executing DigitTimeout(SIP/avenardj-acfc, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(SIP/avenardj-acfc, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround(SIP/avenardj-acfc, demo-congrats) in new stack -- Playing 'demo-congrats' (language 'en') Jun 28 08:41:53 NOTICE[135433216]: sched.c:218 sched_settime: Request to schedule in the past?!?! Jun 28 08:41:56 WARNING[135336960]: chan_sip.c:498 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 25040 (Response) == Spawn extension (default, s, 5) exited non-zero on 'SIP/avenardj-acfc' I'm trying to connect to the SIP gateway over NAT from my home account. Even without NAT when connecting over internet it will not exceed this 5s time limit. It works fine on the local network. I've looked for previous solution and it seems that each time somebody complained about such issue it was related to BSD system. so is asterisk fully working on BSD? If you had this issue in the past ; how did you resolve it? Here is a sample of the sip.conf file for my username: [user1] type=friend nat=yes ; phone may be behing nat host=dynamic reinvite=no canreinvite=no qualify=1000; send udp every now and then to keep nat open mailbox=101 ; mailbox number username=user1 ; username used for identification secret=x ; password for registration dtmfmode=info ; DTMF mode disallow=all allow=gsm allow=ulaw allow=alaw context=sip Also, as a side note. Some people mentioned that they didn't have such issue when the used SER as the SIP proxy ; is it possible to run SER and Asterisk on the same machine? Any ideas? Help please !!! Regards Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA33nxXeDVKqIr3GURAmH/AJ9XAGmv/kubw/HxwtiCn/yajfTo4gCfbyiD ZomB7VHYHeN1d2cc5/4cItQ= =cCFm -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with PostgreSQL
try tcpdump -i lo port 5432 or icmp (or tethereal if you have it) Prehaps it's trying a UNIX socket connection? also, please change your database password as you've now supplied ip,user,pass to the mailing list :) Hopefully, you've got it restricted to localhost Caleb Kow wrote: Here we go: [EMAIL PROTECTED] root]# netstat -ap Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp0 0 *:32768 *:* LISTEN 3221/ tcp0 0 *:imaps *:* LISTEN 3359/couriertcpd tcp0 0 sierra.onefuse.co:32769 *:* LISTEN 3262/xinetd tcp0 0 *:pop3s *:* LISTEN 3380/couriertcpd tcp0 0 *:mysql *:* LISTEN 3326/ tcp0 0 *:poppassd *:* LISTEN 3262/xinetd tcp0 0 *:pop3 *:* LISTEN 3369/couriertcpd tcp0 0 sierra.onefuse.com:783 *:* LISTEN 3598/spamd -d -c -a tcp0 0 *:imap *:* LISTEN 3347/couriertcpd tcp0 0 *:sunrpc*:* LISTEN 3202/ tcp0 0 *:http *:* LISTEN 18983/httpd tcp0 0 *:smtps *:* LISTEN 3262/xinetd tcp0 0 203.208.246.139:domain *:* LISTEN 3273/ tcp0 0 sierra.onefuse.c:domain *:* LISTEN 3273/ tcp0 0 *:ftp *:* LISTEN 3262/xinetd tcp0 0 *:ssh *:* LISTEN 3578/sshd tcp0 0 sierra.onefuse.com:ipp *:* LISTEN 32149/cupsd tcp0 0 *:postgres *:* LISTEN 21845/postmaster tcp0 0 sierra.onefuse.com:rndc *:* LISTEN 3273/ tcp0 0 *:smtp *:* LISTEN 3262/xinetd tcp0 0 *:8443 *:* LISTEN 26729/httpsd tcp0 0 *:https *:* LISTEN 18983/httpd tcp0 20 203.208.246.139:ssh cm55.gamma149.max:29030 ESTABLISHED 25000/sshd tcp0 0 203.208.246.139:ftp cm6.gamma81.maxon:45396 ESTABLISHED 24866/ tcp0 0 203.208.246.139:imapd-105-57.dsl.clea:10824 ESTABLISHED 24740/imapd tcp0 0 203.208.246.139:httpcache51.156ce.max:11750 TIME_WAIT - tcp0 0 203.208.246.139:httpcache51.156ce.max:11621 TIME_WAIT - tcp0 0 203.208.246.139:http216.75.226.2:1763 TIME_WAIT - tcp0 0 203.208.246.139:httpcache51.156ce.max:11746 TIME_WAIT - tcp0 0 203.208.246.139:httpcache51.156ce.max:11745 TIME_WAIT - udp0 0 *:32768 *:* 3221/ udp0 0 *:32769 *:* 3273/ udp0 0 203.208.246.139:domain *:* 3273/ udp0 0 sierra.onefuse.c:domain *:* 3273/ udp0 0 *:853 *:* 3221/ udp0 0 *:sunrpc*:* 3202/ udp0 0 *:631 *:* 32149/cupsd udp0 0 sierra.onefuse.co:43129 sierra.onefuse.co:43129 ESTABLISHED 21845/postmaster On Thu, 24 Jun 2004 17:44:45 -0400, Neil Cherry [EMAIL PROTECTED] wrote: Caleb Kow wrote: Results of netstat -ap You seem to be missing the top part of the output which looks like this: [EMAIL PROTECTED] build]# netstat -ap Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp0 0 *:nfs *:* LISTEN - tcp0 0 *:time *:* LISTEN 3339/xinetd : : (more of the smae looking lines follow). Sorry if that wrapped. Active UNIX domain sockets (servers and established) Proto RefCnt Flags Type State I-Node PID/Program namePath unix 2 [ ACC ] STREAM LISTENING 5881 3623/ /tmp/.iroha_unix/IROHA unix 2 [ ACC ] STREAM LISTENING 3971 3326/ /var/lib/mysql/mysql.sock unix 2 [ ACC ] STREAM LISTENING 6002 3690/ /tmp/jd_sockV4 unix 2 [ ACC ] STREAM LISTENING 9522765 24900/httpd /var/run/fpcgisock
Re: [Asterisk-Users] Call generator
Andrew Kohlsmith wrote: On Wednesday 23 June 2004 04:46, GIBERT Frédéric wrote: Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. Are there any IAX2 call generators? Regards, You can use asterisk to generate the calls, just put a few hundred files in asterisk's spool directory. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 no compatible codecs
check under your network settings that you have all the codecs selected and obviously type IAX Jason Penton wrote: Hi All I have a strange problem using IAX2. When placing a call to my IAX clients (firefly) via the Asterisk dialplan all works great. However trying to initiate a call via the manager interface to the IAX client using the following command results in an error: Action: Originate Channel: IAX2/7000 Extension: 7000 Context: local Priority: 1 ActionID: 1 The error I get in the CLI is Jun 17 08:18:36 WARNING[180236]: chan_iax2.c:4534 socket_read: Call rejected by #IP: No compatible Codecs Does anyone have any ideas. Thanks in advance Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 no compatible codecs
iax2 debug is your friend, looks at the capibilities asterisk is sending in it's NEW message Jason Penton wrote: Hi Adam Done all that but still the same problem. Do you have any other ideas? Cheers Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart Sent: 17 June 2004 08:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 no compatible codecs check under your network settings that you have all the codecs selected and obviously type IAX Jason Penton wrote: Hi All I have a strange problem using IAX2. When placing a call to my IAX clients (firefly) via the Asterisk dialplan all works great. However trying to initiate a call via the manager interface to the IAX client using the following command results in an error: Action: Originate Channel: IAX2/7000 Extension: 7000 Context: local Priority: 1 ActionID: 1 The error I get in the CLI is Jun 17 08:18:36 WARNING[180236]: chan_iax2.c:4534 socket_read: Call rejected by #IP: No compatible Codecs Does anyone have any ideas. Thanks in advance Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another Firefly update - now with SRV support
Kevin P. Fleming wrote: Adam Hart wrote: I've also added support for SIP via TCP and the ability to change the SIP port It complains every time you click OK in the Options page about Changing SIP port requires restart, even if you never looked at the SIP page (and don't even have any SIP networks configured). That's a 'feature' - fixed, new version up ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another Firefly update - now with SRV support
With all the talk of SRV support in Asterisk, I thought I'd add support in Firefly so enjoy. Thanks to Olle for helping me with it, explaining the wonderful world of SIP and SRV to me. There's also an option to disable it (seems to take quite a few DNS lookups for SRV) - warning Duane may hunt you down if you do disable it though :) I've also added support for SIP via TCP and the ability to change the SIP port Yes, it's still version 1.8. Hopefully another little update shortly away too for sip presence. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe have a nice day, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax codec problem
Jason A. Pattie wrote: | | One workaround is to use Firefly, but that may not be for everyone? True. I almost got it working under Wine, though. Kept dumping files into C:\. Probably just means I don't have the necessary dependencies or Wine doesn't have the capabilities needed to run this app., yet. Something about not being able to get timing from threads seemed to be the big killer. fixme:thread:GetThreadTimes Cannot get kerneltime or usertime of other threads fixme:thread:NtQueryInformationThread info class 9 not supported yet Oh well. It was worth a shot. At least part of the interface shows up on the screen before Wine bombs. Lol, that's a decent attempt (funny thing is that's the callstack that's having that problem but I can port it already - just not the GUI) - We're currently looking at porting, looking at the various cross-platform windowing libraries. If you have any suggestions or information on porting a windows GUI C++ program, send me an email cheers, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax codec problem
Tor Houghton wrote: I have the same problem. IAXCOMM works fine with * 0.7.2, but not 0.9. However, you can make calls fine, just not pick up inbound calls. One workaround is to use Firefly, but that may not be for everyone? To the Firefly maintainer: why does the contacts list fill up with copies of calling parties? You mean in the not on list section? - I find it handy to have their numbers on my list, plus it makes it more clear who you're on call with. They'll disappear on restart of Firefly, prehaps I'll include an option to remove on end of call. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
There's a new version out with some bugs fixed major ones fixed: deadlock on call end, iax thread getting locked out, few contact group list bugs, one on exit crash bug fixed I'd highly recommend upgrading to it http://www.virbiage.com/firefly/download/firefly-thirdparty.exe -Adam Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
true, it's internally versioned though - look at the build number. But yes, I'll start suffixing a buildnumber on the files. i'm hoping this will be the last release before the magic feature called conferencing, unless this sip registration issue is firefly related -Adam gARetH baBB wrote: On Wed, 2 Jun 2004, Adam Hart wrote: I'd highly recommend upgrading to it http://www.virbiage.com/firefly/download/firefly-thirdparty.exe Can I recommend you label files with version numbering - this must be about the third ? fourth ? firefly-thirdparty you've released. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
fixed Reto Stauss wrote: Adam The link doesn't seems to work. Get back the following: Parse error: parse error, unexpected T_STRING in /usr/virtual/www.virbiage.com/www/firefly/download/firefly-thirdparty.exe on line 121 Reto There's a new version out with some bugs fixed major ones fixed: deadlock on call end, iax thread getting locked out, few contact group list bugs, one on exit crash bug fixed I'd highly recommend upgrading to it http://www.virbiage.com/firefly/download/firefly-thirdparty.exe -Adam Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
Firefly's RTP port option is for RTP, not SIP listen port. All RTP goes to the one port. There's currently no option to set the SIP port (coming shortly) jo wrote: Adam, works now :-) Just one further question. In my understanding Firefly's RTP Port is the SIP listen port. So there is no chance to influence the RTP/RTCP Portrange for the audio channel. Please correct me if I'm wrong. jo Adam Hart wrote: I just put up another version - fixed that issue and also added to ability to disable registration to a network. Why it's needed? If you will only be making outgoing calls but still need Firefly to use the login info for calling for lazy ppl: http://www.virbiage.com/firefly/download/firefly-thirdparty.exe Quick run down on various ways of calling - 123 - Firefly will find the network marked as internal and dial 123 on that network +123 - Firefly will find the network marked as external and dial 123 (note no plus) on that network. [EMAIL PROTECTED] - Firefly to find the network named blah and dial 123 sip/[EMAIL PROTECTED] (Firefly will try and find the network for this one as well, otherwise make the call as 'guest') (sip:// also works) Otherwise you can use full asterisk urls eg iax/user:[EMAIL PROTECTED]/extension sip/user:[EMAIL PROTECTED]/extension jo wrote: Thanks Adam, no crash after installing over 1.5 B3388. However changing the SIP RTP Port is still not accepted. jo Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
the log looks legit except why does asterisk have a different IP in the contact compared to the 'to' address. I can connect successfully to my asterisk server and FWD - can anyone give me sip access to a asterisk server that firefly doesn't work on? [EMAIL PROTECTED] wrote: Why all the time the firefly show me the message: Sip registration failed for the network Home (407). The server, username and password are correct. I'm using the default RTP port 5000 in the SIP tab. Using the SJPhone I can register; using the firefly I can call any registered number, but I can't register. On asterisk console: Sip read: REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0 To: sip:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]:5060;tag=5a1c4f36 Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060 Expires: 3600 Max-Forwards: 70 User-Agent: Firefly Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.199.121:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=38165263 Content-Length: 0 to 192.168.199.121:5060 SAMPLANET1*CLI Sip read: REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0 To: sip:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060 Expires: 3600 Max-Forwards: 70 Proxy-Authorization: Digest username=2003,realm=asterisk,nonce=38165263,uri=sip:192.168.199.3:5060; transport=udp,response=ec0afc0a2b13a725aa40b5c311c396d8,algorithm=MD5 User-Agent: Firefly Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.199.121:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=38165263 Content-Length: 0 to 192.168.199.121:5060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
get http://www.virbiage.com/firefly/download/g729.zip and follow the instructions (you'll need to compile it) Steven Thomas wrote: adam - can the g729.dll be downloaded somewhere - is this still required for g.729 support? Regards, Steven Thomas *jo [EMAIL PROTECTED]* Sent by: [EMAIL PROTECTED] 31/05/2004 09:19 PM Please respond to asterisk-users To [EMAIL PROTECTED] cc Subject Re: [Asterisk-Users] New Firefly version Thanks Adam, no crash after installing over 1.5 B3388. However changing the SIP RTP Port is still not accepted. jo Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
I'll look at it tomorrow jo wrote: Thanks Adam, no crash after installing over 1.5 B3388. However changing the SIP RTP Port is still not accepted. jo Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly / LibIAX2
It's the standard LibIAX2, the nice features are implemented using text messages. I'd recommend you use the standard LibIAX2 as it's more upto date (Something I've been needing to do too) Reto Stauss wrote: Hi Does anybody know how to build the LibIAX2 from Virbiage? It has some nice features when using Firefly (Messaging, Status Indication). The source can be downloaded here: http://www.virbiage.com/3rdparty/. It does not contain any directions how to compile. Any hints? Thanks! Reto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
I just put up another version - fixed that issue and also added to ability to disable registration to a network. Why it's needed? If you will only be making outgoing calls but still need Firefly to use the login info for calling for lazy ppl: http://www.virbiage.com/firefly/download/firefly-thirdparty.exe Quick run down on various ways of calling - 123 - Firefly will find the network marked as internal and dial 123 on that network +123 - Firefly will find the network marked as external and dial 123 (note no plus) on that network. [EMAIL PROTECTED] - Firefly to find the network named blah and dial 123 sip/[EMAIL PROTECTED] (Firefly will try and find the network for this one as well, otherwise make the call as 'guest') (sip:// also works) Otherwise you can use full asterisk urls eg iax/user:[EMAIL PROTECTED]/extension sip/user:[EMAIL PROTECTED]/extension jo wrote: Thanks Adam, no crash after installing over 1.5 B3388. However changing the SIP RTP Port is still not accepted. jo Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Firefly version
As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
Duane wrote: Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. STUN support doesn't seem to work... Keeps saying unable to contact stun server, and when I did a packet dump and closed and reopened the prog several times I couldn't see any attempts to hit the stun server... STUN server in question (stun.e164.org) works fine with the BT101's... If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| I freshly reinstalled my laptop over the weekend and haven't resinstalled firefly till now... Oops, using a default stun port of 1 - fixing now :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
Just released a minor update http://www.virbiage.com/firefly/download/firefly-thirdparty.exer Fixed STUN - my code was for the old version of STUN RFC. Thanks to Duane for helping debug it. if port 5060 (sip) is in use, it doesn't crash on startup now - just an error message :) I'm guessing this has been a cause of many crashes, people having Xten running in the background. Thanks to Karl for the dump file on that one. keep the bugs coming, Adam PS hope you're enjoying the new contact groups :) Adam Hart wrote: Duane wrote: Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. STUN support doesn't seem to work... Keeps saying unable to contact stun server, and when I did a packet dump and closed and reopened the prog several times I couldn't see any attempts to hit the stun server... STUN server in question (stun.e164.org) works fine with the BT101's... If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| I freshly reinstalled my laptop over the weekend and haven't resinstalled firefly till now... Oops, using a default stun port of 1 - fixing now :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: FireFly doesn't work with 3rd party anymore
I'm going to have to go against this statement, there's one bug that I need to fix so unfortunately it will have to be Monday now. For those after the IAX/SIP firefly (albeit an old version) get http://www.virbiage.com/firefly/download/firefly-dev.exe apologies, Adam Adam Hart wrote: They'll be a new version at the end of the day (it's 9:25am now) - The reason it was like that was to cope with overlap for the firefly network going to Freshtel. Freshtel will have the Firefly Network and special version of Firefly (no IAX and SIP) while Virbiage will have a standard IAX and SIP client. Freshtel has taken our Firefly Network to allow us to concentrate on Hardware (Insert vaporware joke here) If anyone's after Australian IAX termination (or Australians wishing to call overseas), try www.freshtel.net - iax server is ctsau.freshtel.net sorry for the dodgy version, Adam usedcanon wrote: Quite interesting, since there version history say 1.4 is the latest. The one you download is 1.7 and only works with Firefly. I have V1.5 which has the option to connect to other services. I am interested to know whats the highest version anyone has that has the other services options. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: 27 May 2004 19:30 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: FireFly doesn't work with 3rd party anymore In article [EMAIL PROTECTED], I wrote: In article [EMAIL PROTECTED], brian [EMAIL PROTECTED] wrote: Just an FYI FireFly no longer works with anything but the FireFly network. No more SIP, No more IAX. It was a damn good IAX client... too bad its crap now. Are you sure? http://www.virbiage.com/firefly/download/ still says the following: Standalone SIP / IAX mode: If you want to use Firefly on our Firefly phone network (with your own voicemail etc.) then you will need to register a phone number. However, you can also use Firefly as a SIP or IAX client on your own network. Well, I just downloaded the new 1.7 build from their website (from the same page that states the above), and I see what you mean. When I first ran the new version, it still used my old settings, and successfully connected to my Asterisk server. I looked in the Options dialog, and as you say, there is no third party option at all, only the option to connect to the Firefly network. Moreover, when I changed an unrelated option (sound output device), it then overwrote my settings in the registry with new settings for the Firefly network, Freshtel. Not impressed. Especially since in their FAQ they still explicitly say it can be used with Asterisk systems. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: FireFly doesn't work with 3rd party anymore
They'll be a new version at the end of the day (it's 9:25am now) - The reason it was like that was to cope with overlap for the firefly network going to Freshtel. Freshtel will have the Firefly Network and special version of Firefly (no IAX and SIP) while Virbiage will have a standard IAX and SIP client. Freshtel has taken our Firefly Network to allow us to concentrate on Hardware (Insert vaporware joke here) If anyone's after Australian IAX termination (or Australians wishing to call overseas), try www.freshtel.net - iax server is ctsau.freshtel.net sorry for the dodgy version, Adam usedcanon wrote: Quite interesting, since there version history say 1.4 is the latest. The one you download is 1.7 and only works with Firefly. I have V1.5 which has the option to connect to other services. I am interested to know whats the highest version anyone has that has the other services options. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: 27 May 2004 19:30 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: FireFly doesn't work with 3rd party anymore In article [EMAIL PROTECTED], I wrote: In article [EMAIL PROTECTED], brian [EMAIL PROTECTED] wrote: Just an FYI FireFly no longer works with anything but the FireFly network. No more SIP, No more IAX. It was a damn good IAX client... too bad its crap now. Are you sure? http://www.virbiage.com/firefly/download/ still says the following: Standalone SIP / IAX mode: If you want to use Firefly on our Firefly phone network (with your own voicemail etc.) then you will need to register a phone number. However, you can also use Firefly as a SIP or IAX client on your own network. Well, I just downloaded the new 1.7 build from their website (from the same page that states the above), and I see what you mean. When I first ran the new version, it still used my old settings, and successfully connected to my Asterisk server. I looked in the Options dialog, and as you say, there is no third party option at all, only the option to connect to the Firefly network. Moreover, when I changed an unrelated option (sound output device), it then overwrote my settings in the registry with new settings for the Firefly network, Freshtel. Not impressed. Especially since in their FAQ they still explicitly say it can be used with Asterisk systems. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: FireFly doesn't work with 3rd party anymore
Adam Goryachev wrote: On Fri, 2004-05-28 at 09:28, Adam Hart wrote: If anyone's after Australian IAX termination (or Australians wishing to call overseas), try www.freshtel.net - iax server is ctsau.freshtel.net Except I get: [EMAIL PROTECTED]: ~$ mtr ctsau.freshtel.net mtr: Unknown host Perhaps you could just let people know what connectivity options you have (ie, what your IP interconnect point are...) Currently I would have to cross telstra + CCA... Regards, Adam cts-au.freshtel.net sorry, it's hosted at comindico in sydney. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: FireFly doesn't work with 3rd party anymore
Adam Goryachev wrote: I suppose I could do QoS on outbound, which should improve things somewhat for the remote caller, but that doesn't help inbound packets. Does anyone have any comments on what this would mean for VoIP calls with the above variables? I think the biggest problem is the jitter, does the IAX jitter buffer work at the moment? Would it keep things working reasonably under the above circumstances? Regards, Adam Depends on your end client, a Voip phone will handle that fine, otherwise I have no idea regarding the quality of the IAX jitter buffer - try it and see :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy with kernel 2.6
Tony Hoyle wrote: Scott Brooks wrote: Has anyone ported the ztdummy module to 2.6? I don't really want to dive into it that far if someone already has. http://www.nodomain.org/asterisk/ztdummy.diff :) Tony Forgive me for prehaps a stupid question but does the 2.6 kernel have accurate timers built in now? as I see your code just wraps their timer. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy with kernel 2.6
Tony Hoyle wrote: Adam Hart wrote: Forgive me for prehaps a stupid question but does the 2.6 kernel have accurate timers built in now? as I see your code just wraps their timer. The HZ value in 2.6 is now 1000 to support realtime scheduling etc. It's certainly accurate enough. I'm not sure the same thing would work on 2.4 though (maybe worth a try if someone's got some spare time...) If you run zttest with it loaded you get about 99.98% accuracy. Tony Great news then, prehaps there's a way to interface directly from user mode so drivers ain't required at all (fallback mode if no zaptel hardware installed) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NetJet and RAS
Andrew Yager wrote: Hi, Last weekend I was planning to buy a physical PBX system, but instead I have been blown away by the fact that VoIP really works, that Asterisk is so easy to set up and use... and free! We're in Australia, so as I understand it, we aren't allowed to use the Zaptel cards. By not allowed, you mean not Austel approved - then yes, you shouldn't use it but it will still work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 codec for asterisk
Kevin Walsh wrote: brian [EMAIL PROTECTED] wrote: I've seen that licenses are purchased on a per-channel basis. Could we make some sort of agreement on having a no-limit channel license? Even, we would like to have the possibility of installing it on how many machines we wish to do. No you MUST pay per channel because the patent holders require that. The patent holders would [EMAIL PROTECTED] kittens if you had no port limit or any type of control on it. That's why the control and registration processes are in place to comply with the patent holders requirements. So your request translates into I want something for nothing. I see it as the patent holders who want something for nothing. Haven't they been paid enough millions to justify that tiny amount of work yet? CELP (Code Excited Linear Prediction) and (more specifically) ACELP was a revolutionary change in encoding of voice, it ain't a 'tiny amount of work'. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] speex
compared to? My P4/Xeon 2.8 does SLINR - iLBC in 12ms so a 2.4ghz should take 14 (?) Andrew Kohlsmith wrote: -fPIC -O3 -march=pentium4 -funroll-loops -fomit-frame-pointer -msse -msse2 -mfpmath=sse Made no difference whatsoever on my P4/Xeon 2.4GHz for iLBC (I don't use speex). -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] speex
Actually it encodes a second of data, which with a 20ms codec would be 50 frames. The timing shows better than expected results due to caching. -Adam brian wrote: http://asterisk.bkw.org/diff/translate.patch.txt If you try that patch out it adds a nice feature... show translation recalc [xx] You can throw more than 1 sample thru it and recalculate your translation matrix. It also allows you to see TRUE translation under a load or just when ever you feel like seeing them updated. When * loads the codec it shoot one frame thru and times it. Now under real world scenarios you will be shooting more than one frame thru so LETS have the option to update the matrix with these types of tests. 200 is the max. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James H. Cloos Jr. Sent: Monday, May 17, 2004 11:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] speex Just a suggestion to anyone using speex: Try running the 1.1.5 or svn code rather than 1.0.3. As a quick example, here are the show translation outputs from * on a 2.8 GHz P4 with speex 1.0.3 (from debian sid's .deb) and from 1.1.5 (compiled with CFLAGS=-march=pentium4 and --enable-sse). Note how encoding from slin went from 25 to 15 ms. That is from the re-write of the sse optimized routines in libspeex. The % change is similar to what I saw on my p3 notebook, where both 1.0.3 and 1.1.5 were compiled with --enable-sse and -march=pentium3. (As a side note, these were captured before Brian's ilbc Makefile patch made it to the anon cvs tree; that optimization shaved 5ms off the time to encode to iLBC on that box.) spx103*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -2619 ULAW - 3 - 1 2 2 1 6 -2619 ALAW - 3 1 - 2 2 1 6 -2619 G726 - 3 2 2 - 2 1 6 -2619 ADPCM - 3 2 2 2 - 1 6 -2619 SLINR - 2 1 1 1 1 - 5 -2518 LPC10 - 4 3 3 3 3 2 - -2720 G729A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - -19 ILBC - 5 4 4 4 4 3 8 -28 - spx115*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -1619 ULAW - 3 - 1 2 2 1 6 -1619 ALAW - 3 1 - 2 2 1 6 -1619 G726 - 3 2 2 - 2 1 6 -1619 ADPCM - 3 2 2 2 - 1 6 -1619 SLINR - 2 1 1 1 1 - 5 -1518 LPC10 - 4 3 3 3 3 2 - -1720 G729A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - -19 ILBC - 5 4 4 4 4 3 8 -18 - -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com/voip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] speex
Btw, Good work. 5ms is a huge different, espically in optimizing terms. I've added a few flags and shaved off another ms here's my flags: (only for p4/xeon) -fPIC -O3 -march=pentium4 -funroll-loops -fomit-frame-pointer -msse -msse2 -mfpmath=sse keep up the good work, Adam brian k. west wrote: Yes I realized my error in my wording but it was early :P It doesn't improve alot but does give you some ways to get a better idea of translation times if your box is loaded up with calls. bkw PS this patch was added to CVS-HEAD - Original Message - From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 5:49 PM Subject: Re: [Asterisk-Users] speex Actually it encodes a second of data, which with a 20ms codec would be 50 frames. The timing shows better than expected results due to caching. -Adam brian wrote: http://asterisk.bkw.org/diff/translate.patch.txt If you try that patch out it adds a nice feature... show translation recalc [xx] You can throw more than 1 sample thru it and recalculate your translation matrix. It also allows you to see TRUE translation under a load or just when ever you feel like seeing them updated. When * loads the codec it shoot one frame thru and times it. Now under real world scenarios you will be shooting more than one frame thru so LETS have the option to update the matrix with these types of tests. 200 is the max. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James H. Cloos Jr. Sent: Monday, May 17, 2004 11:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] speex Just a suggestion to anyone using speex: Try running the 1.1.5 or svn code rather than 1.0.3. As a quick example, here are the show translation outputs from * on a 2.8 GHz P4 with speex 1.0.3 (from debian sid's .deb) and from 1.1.5 (compiled with CFLAGS=-march=pentium4 and --enable-sse). Note how encoding from slin went from 25 to 15 ms. That is from the re-write of the sse optimized routines in libspeex. The % change is similar to what I saw on my p3 notebook, where both 1.0.3 and 1.1.5 were compiled with --enable-sse and -march=pentium3. (As a side note, these were captured before Brian's ilbc Makefile patch made it to the anon cvs tree; that optimization shaved 5ms off the time to encode to iLBC on that box.) spx103*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -26 19 ULAW - 3 - 1 2 2 1 6 -26 19 ALAW - 3 1 - 2 2 1 6 -26 19 G726 - 3 2 2 - 2 1 6 -26 19 ADPCM - 3 2 2 2 - 1 6 -26 19 SLINR - 2 1 1 1 1 - 5 -25 18 LPC10 - 4 3 3 3 3 2 - -27 20 9A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - - 19 ILBC - 5 4 4 4 4 3 8 - 8 - spx115*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -16 19 ULAW - 3 - 1 2 2 1 6 -16 19 ALAW - 3 1 - 2 2 1 6 -16 19 G726 - 3 2 2 - 2 1 6 -16 19 ADPCM - 3 2 2 2 - 1 6 -16 19 SLINR - 2 1 1 1 1 - 5 -15 18 LPC10 - 4 3 3 3 3 2 - -17 20 9A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - - 19 ILBC - 5 4 4 4 4 3 8 - 8 - -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com/voip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo
Re: [Asterisk-Users] iax behind a SonicWall
John Todd wrote: At 8:23 PM -0600 on 5/12/04, Rich Adamson wrote: Current dev cvs install on two systems. System A is behind a SonicWall firewall, and system B is on a registered IP address. (System B has multiple iax links that are fully functional to multiple locations.) System A is correctly registering with System B, with no special firewall rules. Should System B be able to take advantage of the registration to send iax/gsm calls to System A without installing any firewall rules? I assumed it could, but an ethereal trace indicates a new call is placed from B - A, but A never acknowledges the iax2 packet, etc. The trace suggests the registration is happening with src port 28277 (or whatever) - dest port 4569 however, calls are processed with src port 4569 and dest port 4569 Shouldn't we expect src=4569 and dest=4569 on all iax2 interactions? Rich If src=4569 and dst=4569 always, then it would be impossible to have more than one IAX2 talker behind a firewall talking to an external Asterisk server, right? There would be no method by which the firewall would know which packet was destined for what device inside the firewall, since the source port and destination port would be the same for each connection. I'm not thinking this through completely, and it seems like there is a flaw in this argument... but with UDP, there is no sequence number that should have attention paid to it (like TCP) so... er... someone tell me why this is incorrect. note: firewall in this case is really NAT, right? Correct, a NAT will allocate a unique port for each talker so it's very common that you'll see connections coming from random ports. Even more sexy, if the NAT follows the RFC (can't remember the number) suggestion it will reuse the same port if the packets are going to another ip. example 192.168.0.3 (A) wants to reach 1.2.3.4 port 4569, NAT allocates port 1234 for it 192.168.0.2 (B) wants to reach 1.2.3.4 port 4569, NAT allocates port for it 192.168.0.3 (A) wants to reach 4.3.2.1 port 4569, NAT reuses port 1234 for it. This is essensial for native transfers for NAT to NAT as 1.2.3.4 (in this case) can predict what port the NAT will allocate for the transfer. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Virbiage FT201 IAX Hard Phone
We're waiting on the processor chip to be made for our first production run, there's currently no stock and they're in the process of making more. It's completely out of our hands and, trust me, I'm as frustrated as you guys are. As soon as our manufactures tell us the completion date, I'll post. -Adam Brian D'Arcy wrote: Does anyone have any recent news on the Virbiage FT201 IAX Hardphone? I'd *really really* like to deploy these phones instead of SIP hardphones, and I can't help but wonder if I'm going to shoot myself in the foot (or another sensitive area) by deploying a ton of SIP phones just to find the IAX Hardphones were released a week later... Thanks, Brian D'Arcy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323, compile problems using V0.6.0 or 0.6.1
apply the openh323 patch (it's in the root of ast-oh323), recompile
openh323 and it should work fine
David Hindmarsh wrote:
Hi
I have recently updates to the latest cvs of asterisk, openh323 and pwlib as recommended.
The OPenh323 and pwlib compile fine.
When compiling the Asterisk-oh323 I get the following errors, I have checked that the envorinment variables are set correctlty as below.
PWLIBDIR=/usr/src/pwlib
OPENH323DIR=/usr/src/openh323
LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib
g++ (GCC) 3.3.1 (SuSE Linux)
The errors from the compile are below
mipt:/usr/src/asterisk-oh323-0.6.1 # make
for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
make: *** No rule to make target `ccflags'. Stop.
make: *** No rule to make target `ccflags'. Stop.
make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.1/wrapper'
./check_ver /usr/src/pwlib pwlib
./check_ver /usr/src/openh323 openh323
g++ -Wall -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.7.0\
-DOPENH323VERSION=\1.14.0\ -I/usr/include/openssl -I/usr/src/pwlib/include/ptlib/unix
-I/usr/src/pwlib/include -I/usr/src/openh323/include -I/usr/src/openh323/include/openh323
-I../asterisk-driver -x c++ -Os -g -c wrapper_misc.cxx -o wrapper_misc.o
In file included from /usr/src/pwlib/include/ptlib.h:172,
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
/usr/src/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: error: parse error before
`protected'
/usr/src/pwlib/include/ptlib/unix/ptlib/pdirect.h:80: error: syntax error
before `*' token
In file included from /usr/src/pwlib/include/ptlib.h:184,
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
/usr/src/pwlib/include/ptlib/unix/ptlib/config.h:53: error: parse error before
`public'
/usr/src/pwlib/include/ptlib/unix/ptlib/config.h:55: error: destructors must be
member functions
/usr/src/pwlib/include/ptlib/unix/ptlib/config.h:57: error: parse error before
`protected'
In file included from /usr/src/pwlib/include/ptlib.h:190,
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
/usr/src/pwlib/include/ptlib/args.h:121: error: parse error before `{' token
/usr/src/pwlib/include/ptlib/args.h:147: error: parse error before `const'
/usr/src/pwlib/include/ptlib/args.h:156: error: parse error before `const'
/usr/src/pwlib/include/ptlib/args.h:165: error: parse error before `int'
/usr/src/pwlib/include/ptlib/args.h:175: error: parse error before `int'
/usr/src/pwlib/include/ptlib/args.h:190: error: `ostream' was not declared in
this scope
/usr/src/pwlib/include/ptlib/args.h:191: error: `strm' was not declared in this
scope
/usr/src/pwlib/include/ptlib/args.h:191: error: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:191: error: variable or field `PrintOn'
declared void
/usr/src/pwlib/include/ptlib/args.h:197: error: `istream' was not declared in
this scope
/usr/src/pwlib/include/ptlib/args.h:198: error: `strm' was not declared in this
scope
/usr/src/pwlib/include/ptlib/args.h:198: error: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:198: error: variable or field `ReadFrom'
declared void
/usr/src/pwlib/include/ptlib/args.h:206: error: parse error before `' token
/usr/src/pwlib/include/ptlib/args.h:215: error: parse error before `' token
/usr/src/pwlib/include/ptlib/args.h:246: error: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:249: error: parse error before `' token
/usr/src/pwlib/include/ptlib/args.h:254: error: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:266: error: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:266: error: non-member function `PINDEX
GetOptionCount(char)' cannot have `const' method qualifier
/usr/src/pwlib/include/ptlib/args.h:270: error: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:270: error: non-member function `PINDEX
GetOptionCount(const char*)' cannot have `const' method qualifier
/usr/src/pwlib/include/ptlib/args.h:273: error: parse error before `' token
/usr/src/pwlib/include/ptlib/args.h:274: error: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:274: error: non-member function `PINDEX
GetOptionCount(...)' cannot have `const' method qualifier
/usr/src/pwlib/include/ptlib/args.h:283: error: non-member function `BOOL
HasOption(char)' cannot have `const' method qualifier
/usr/src/pwlib/include/ptlib/args.h:287: error: non-member function `BOOL
HasOption(const char*)' cannot have `const' method qualifier
/usr/src/pwlib/include/ptlib/args.h:290: error: parse error before `' token
/usr/src/pwlib/include/ptlib/args.h:291: error: non-member function `BOOL
HasOption(...)' cannot have `const' method qualifier
/usr/src/pwlib/include/ptlib/args.h:301: error: syntax error before `(' token
/usr/src/pwlib/include/ptlib/args.h:306:
Re: [Asterisk-Users] G.723
rr80 wrote: Is there is support for G.723 codec in Asterisk 0.7.2+Astrisk-OH323 0.5.10 or it should be bought separately like G.729? - Pavel Riko ___ Neither, you can't get asterisk to en/decode G.723.1 - only proxy it ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence
Duane wrote: William Suffill wrote: They modified iax to include the presence packet but only works on their customized firefly network. I was thinking along the lines of a software app for those of us who use hardware phones but still want to keep TXT chat and presence and perhaps integrated into 1 of the iax soft phones as well to provide a full solution. Question is then, how well does their system work? Already have an IAX2 compatible soft phone with that stuff in it, why not make use of the fact and just work out what needs to be sent to their client... The protocol is quite simple, it's all text messages. S for subscribe to a user's events, T for send a text message I was half way through discussing this with Mark and more specifically adding it to IAX (along with some other cool stuff). Unforunately, I was told to do another project asap but that'll be released next week (stay tuned). My main concern with IAX were you don't know when someone goes offline until their reg expires - no acceptable in presence. Our solution was to keep the registration session open. Keeping the registration session open actually helps everything else fall in place, you can just send messages over that session without requiring setting up a channel, auth and tear down for each message. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Capacity
WipeOut wrote: Doesn't NuFone use SER in front of Asterisk? so using asterisk purely as the PSTN gateway.. Later Nufone offers IAX termination, SER is SIP - or am I missing something here? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly Client can't receive incoming calls
Ken DeMaria wrote: I'm having a problem configuring asterisk to send incoming calls to Firefly.I can make outgoing calls from firefly through asterisk without any problems at all. The firefly client does this when it's on the same IP subnet without a firewall, or from a NAT'd environment. Can anyone tell me where I'm going wrong? do a iax2 debug - should help you diagnose the issue ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Virbiage Phones - Vapourware??
Aaron Martin wrote: Has anyone heard any more info about the Virbiage FT201 VoIP phones? About 3 months ago I was told they were 6 weeks away, about 3 weeks ago I was told they were 2 weeks away, and now I am told they are 2 months away again! Are they EVER going to arrive? Can anyone shed some light on this? I'm not sure who's been telling you 2 weeks away but I've posted previously that the phones were 8 weeks away. The hardware is completely done, I have one on my desk, software will be finished soon and we are mainly just waiting on some parts for the first run. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 in Asterisk
Read README under channels/h323, it should point you in the right direction Terence Parker wrote: I have posted before but didn't get any replies so i'll ask again in a more simple way : Does H323 work on asterisk out of the box? I notice there is already a channels/chan_h323.c file, but creating an h323.conf file I can't seem to get H323 working. Do I have to compile an additional package first or something? I tried the asterisk-oh323 thing, but can't get it to compile. Terence ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 variants and Asterisk
Carlos Chavez wrote: I see that I can purchase G.729 licenses for my Asterisk server, but I have seen that many phones support a G.729 variant like A or B. Are these suppoted by the same G.729 codec in Asterisk? B is just the fixed point version of A (from memory) - so it works the same as A. A is a reduced complexity version of G.729 - although they both work with each other. A is just slack when looking for the best representation of your voice. FYI, Digium's codec is G.729A, although it makes little difference ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New minor release of Firefly (now with Speex)
I've put up a new dev version of Firefly (http://www.virbiage.com/firefly/download/firefly-dev.exe) Notable Changes: DTMF now works with SIP Speex codec has been added 1 crash bug fixed - 2 more to go (if you can crash Firefly, send me the Hex address - probably stored in event viewer under control panel) Sorry for the delay but I've completely rewritten how contacts work internally (although it looks exactly the same as it did before). This now allows me to do some sexy things with contacts. Stay tuned I'm aiming for a stable release in two weeks so help me find the bugs. Many thanks to thoses who have -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 as an IETF Standard?
Olle E. Johansson wrote: An informational RFC documenting the protocol would be a good start, it would make it more open but not an IETF product. Security specialists would get something to read and analyze. A VOIP protocol with RSA authentication, implemented today. Is there any IAX2 document that could be a basis document somewhere? Someone has written a IAX2 document. It's on the mailing list.. somewhere IMO, IAX2 needs some more works before it's finialized. I'm feeling very guilty as I promised Mark I'd implement some encryption stuff but haven't got around to it. (AES voice encryption baby) I'd like to see some of firefly's features in IAX2 as well. I also like to see two people behind the same nat being able to communicate directly (without requiring pin-wheeling). Ie The client attaches their private ip to the register packet, which is used when client A B's public ips match. Once I release the new firefly, I'll get back to it. Sorry Mark -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 as an IETF Standard?
Robert Hajime Lanning wrote: quote who=Adam Hart I also like to see two people behind the same nat being able to communicate directly (without requiring pin-wheeling). Ie The client attaches their private ip to the register packet, which is used when client A B's public ips match. 192.168.1.0/24 -- NAT-BOX -- Internet -- NAT-BOX -- 192.168.1.0/24 | | | IAX phoneAsterisk-Box IAX phone umm... I would suggest the default setting to be off, as the above topology would be very common. from my post: which is used when client A B's public ips match. meaning in this situation both clients would have different public IPs and it wouldn't be used. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 as an IETF Standard?
James H. Thompson wrote: No guarantee then when public IPs match that clients are both on same NAT LAN. Client A 192.168.0.1 - NAT Router A - NAT Router X with Public IP 123.123.123.123 --- Internet Client B 192.168.0.1 - NAT Router B -| Jim James H. Thompson [EMAIL PROTECTED] Very true, solution would be try both. If private fails, try public ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 as an IETF Standard?
Comment below... Steve wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 24 March 2004 08:45 pm, James H. Thompson wrote: No guarantee then when public IPs match that clients are both on same NAT LAN. Client A 192.168.0.1 - NAT Router A - NAT Router X with Public IP 123.123.123.123 --- Internet Client B 192.168.0.1 - NAT Router B -| The thing is that it's all controlled by your gateway configuration. This is where you define where you find what. You must know the IP (or domain name and use DNS) of where the recipient is. If you are calling a local host you must know the IP. If you call an external host you must also must know his internet address. He'd have a redirect in his firewall that would route to his internal machine. You have no need/use of knowing what his internal IP address is. I've done all the above in many combinations. I have one setup on CA and one in FL. I have had CA call over IP to FL, then fwd the call to a local external land line and call right back in again on another land line. I have called and transferred calls to a local LAN phone as well as over the Internet. I can't really follow what you're saying, the above setup is a problem with the current IAX. Put simply, when two people are behind the same NAT device and the asterisk box is outside this nat, some NAT routers can't bridge the calls so the call is forced to continue to route through the asterisk box. This is most common cause of compliant of latency for the firefly network. Sure SOME routers understand but most don't. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] firefly softphone
Simon Brown wrote: I had exactly the same problem. I tried removing and reinstalling several times but it always crashed. I sent an email to verbiage asking for help and all I got in response was Have you got it working yet? from them. I have been unable to get a reply since. Simon Brown Are you using http://www.virbiage.com/firefly/download/firefly-dev.exe If possible, could you get the Hex address, which XP stores under Event Viewer (in admin tools) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The FT201 is currently being manufactured and will be available shortly! The retail price will be $129.95 USD
Dave Cotton wrote: On Wed, 2004-03-17 at 04:43, Adam Hart wrote: Eric Wieling wrote: 6) are there USA resellers Yes, many USA resellers have expressed interest. Virbiage won't be selling directly. And the 255 million people in Europe? Please not the usual, 75US$ for the unit 80US$ for FedEx or UPS to deliver it from the US. Of course there will be resellers in Europe, Eric asked about USA resellers. Basically, if you want to resell, you can. We have had requests from all over the world. We'll be looking more closely at reseller agreements closer to the launch (we'll send them samples and such) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] firefly sip question
hank smith wrote: hello I am not sure where to ask this question at so please except my apologise if this is the wrong list. I need to ask if any one has got firefly sip version to work with fre world dialup? if so what info did they use to connect? once again if this is the wrong list if the person who is developing this thing email me off list or direct me to a list fore firefly it would be greatly appreciated email is [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] thanks hank I'll look at this today, stay tuned. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly Beta - with SIP and G.729
Jim Flagg wrote: Firefly's Protocol Support now is: Voip Protocols: SIP, IAX Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL) Sounds good. Any plans for Speex codec support? ___ Adding it this week, along with some bug fixes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The FT201 is currently being manufactured and will be available shortly! The retail price will be $129.95 USD
Eric Wieling wrote: The FT201 is currently being manufactured and will be available shortly! The retail price will be $129.95 USD... http://www.virbiage.com/products/lanphones.php The web page does not say: 1) how many call appearances does the phone has It can present 5 calls and you can action each via its button. It can also handle multiple calls and conference them together. Or have I misunderstood what you mean? 2) does firmware costs extra Everything's included 3) does it come with a power supply Of course 4) does it support PoE. The first batch of phones won't but the board's ready for it (just needs the chip). You can be assured a PoE version will be available 5) are all the features listed available in the initial release of the firmware Yes (it will use a modified firefly call stack) 6) are there USA resellers Yes, many USA resellers have expressed interest. Virbiage won't be selling directly. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The FT201 is currently being manufactured and will be available shortly! The retail price will be $129.95 USD
Matthew Marlowe wrote: (reposted to be in text format, sorry. :)) The FT201 is currently being manufactured and will be available shortly! The retail price will be $129.95 USD... http://www.virbiage.com/products/lanphones.php Let me clarify the FT 201 situation, the current ETA is 8 weeks. The main delay is on a few parts. Although the retail price is $129, I'm guessing resellers will be selling under that (just based on how much the BudgeTone is selling for vs retail). Software progress is going well, it runs linux and firefly's call stack (modified as most is done via the DSP). We'll be asking the community soon their wishlist :) -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly Beta - with SIP and G.729
Just a quick update, there's was a problem with SIP - if you were getting SIP registration failed, grab the new version. (http://www.virbiage.com/firefly/download/firefly-dev.exe) thanks for the feedback about this bug, Adam Adam Hart wrote: I've been sitting on this release for a week so I thought I'd better just release it :) Firefly now has SIP but it's still in a beta state. If you manage to crash it, send me the hex address of the crash. If you find it doesn't work with another SIP phone, let me know and I'll happy get it working for you. I'll be interested to hear people's experiences behind NATs. To download the beta version of Firefly: http://www.virbiage.com/firefly/download/firefly-dev.exe (the current stable version of firefly will not have sip or g.729) G729 support via dll - basically as we all know, G.729 ain't free but you can get a free development version from Voiceage (Sipro), so I've added support for using that. Download http://www.virbiage.com/firefly/download/g729.zip and follow the instructions in the Readme. You'll need to agree to their license and download their library. Firefly's Protocol Support now is: Voip Protocols: SIP, IAX Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL) Next major feature will be conferencing. feel free to email me, Adam Hart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly Beta - with SIP and G.729
I'll look at it tomorrow, what url are you using? standard asterisk syntax? Stig Andersson wrote: Hi again, Installed your new release today (after the sip bugfix). Now SIP registers OK with asterisk, but calling fails... Firefly says: Couldn't start call. Asterisk in SIP debug mode shows the registration, but shows no response when firefly tries to call. Using NO stun, asterisk and Firefly on the same net, using only code G:711 u/alaw Registration data follows if of interrest... Regards Stig - Sip read: REGISTER sip:asterisk.ymex.com:5060;transport=udp SIP/2.0 To: sip:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]:5060;tag=014ee749 Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-386924776-1--c87542-;rport Call-ID: c75e00726c471711 CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060 Expires: 3600 Max-Forwards: 70 User-Agent: Firefly Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 217.119.162.35 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-386924776-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=014ee749 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 217.119.162.35:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-386924776-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=014ee749 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=30bc622a Content-Length: 0 to 217.119.162.35:5060 asterisk*CLI Sip read: REGISTER sip:asterisk.ymex.com:5060;transport=udp SIP/2.0 To: sip:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport Call-ID: c75e00726c471711 CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060 Expires: 3600 Max-Forwards: 70 Proxy-Authorization: Digest username=stig,realm=asterisk,nonce=30bc622a,uri=sip:asterisk.ymex.com:5060;transport=udp,response=d39488505ce4c15723e4b8f3a7a2bb69,algorithm=MD5 User-Agent: Firefly Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 217.119.162.35 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 217.119.162.35:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED];expires=3600 Date: Wed, 17 Mar 2004 07:24:46 GMT Content-Length: 0 to 217.119.162.35:5060 At 17:34 2004-03-17 +1100, you wrote: Just a quick update, there's was a problem with SIP - if you were getting SIP registration failed, grab the new version. (http://www.virbiage.com/firefly/download/firefly-dev.exe) thanks for the feedback about this bug, Adam Adam Hart wrote: I've been sitting on this release for a week so I thought I'd better just release it :) Firefly now has SIP but it's still in a beta state. If you manage to crash it, send me the hex address of the crash. If you find it doesn't work with another SIP phone, let me know and I'll happy get it working for you. I'll be interested to hear people's experiences behind NATs. To download the beta version of Firefly: http://www.virbiage.com/firefly/download/firefly-dev.exe (the current stable version of firefly will not have sip or g.729) G729 support via dll - basically as we all know, G.729 ain't free but you can get a free development version from Voiceage (Sipro), so I've added support for using that. Download http://www.virbiage.com/firefly/download/g729.zip and follow the instructions in the Readme. You'll need to agree to their license and download their library. Firefly's Protocol Support now is: Voip Protocols: SIP, IAX Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL) Next major feature will be conferencing. feel free to email me, Adam Hart
[Asterisk-Users] New Firefly Beta - with SIP and G.729
I've been sitting on this release for a week so I thought I'd better just release it :) Firefly now has SIP but it's still in a beta state. If you manage to crash it, send me the hex address of the crash. If you find it doesn't work with another SIP phone, let me know and I'll happy get it working for you. I'll be interested to hear people's experiences behind NATs. To download the beta version of Firefly: http://www.virbiage.com/firefly/download/firefly-dev.exe (the current stable version of firefly will not have sip or g.729) G729 support via dll - basically as we all know, G.729 ain't free but you can get a free development version from Voiceage (Sipro), so I've added support for using that. Download http://www.virbiage.com/firefly/download/g729.zip and follow the instructions in the Readme. You'll need to agree to their license and download their library. Firefly's Protocol Support now is: Voip Protocols: SIP, IAX Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL) Next major feature will be conferencing. feel free to email me, Adam Hart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
