[asterisk-users] From Domain in REGISTER string
Hi Below is my register string register => usern...@test.abc.com:xxx:uern...@test.server.com The REGISTER from asterisk has the From header with test.server.com instead of test.abc.com Any help appreciated. UK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] REINVITE with Auth Credentials has different SDP Codec
What am I doing wrong...to get no responses at all Thx From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo Sent: Thursday, October 07, 2010 7:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] REINVITE with Auth Credentials has different SDP Codec Hi I have a call from Service Provider (SP) to Asterisk to User User sends a T38 REINVITE Asterisk passes that to SP SP challenges the INVITE Asterisk sends INVITE with credentials but sends G711ulaw in the SDP instead of T38 udptl... Obviously Fax fails.. Any ideas on how I can maintain the T38 SDP when SP challenges the mid-Call T38 REINVITE? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] REINVITE with Auth Credentials has different SDP Codec
Hi I have a call from Service Provider (SP) to Asterisk to User User sends a T38 REINVITE Asterisk passes that to SP SP challenges the INVITE Asterisk sends INVITE with credentials but sends G711ulaw in the SDP instead of T38 udptl... Obviously Fax fails.. Any ideas on how I can maintain the T38 SDP when SP challenges the mid-Call T38 REINVITE? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registering Multiple Trunks to Service Provider
Any pointers on this one?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo
Sent: Monday, October 04, 2010 12:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Registering Multiple Trunks to Service Provider
We have multiple entries like the one below in our users.conf file... where the
username. Contact and secret changes for different customers and we register on
their behalf to the Service Provider.
For the trunk below: when the call is placed out, Asterisk (1.4.18) sends the
username of "abc.com" in the MD5 Auth .which obviously does not match the
trunk setup for this Customer with our Service Provider (username below is
3035551122)
I don't see anywhere any config file the username = abc.com where could the
asterisk be picking it up from?
We have more than 10 such entries (all with same host = provider.sip.com value)
and when as INVITE is challenged, the Asterisk does match the correct trunk and
seems to send out correct Auth credentials...but not the one below..
[trunk_1]
;register to SP
allow = ulaw
;context = test
dialformat = ${EXTEN:1}
canreinvite = no
hasexten = no
hasiax = no
hassip = yes
host = provider.sip.com
insecure = very
port = 5060
registeriax = no
registersip = yes
trunkname = test
trunkstyle = customvoip
username = 3035551122
disallow = gsm,g726,alaw
contact = 3035551122
secret = x
--
_
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[asterisk-users] Registering Multiple Trunks to Service Provider
We have multiple entries like the one below in our users.conf file... where the
username. Contact and secret changes for different customers and we register on
their behalf to the Service Provider.
For the trunk below: when the call is placed out, Asterisk (1.4.18) sends the
username of "abc.com" in the MD5 Auth .which obviously does not match the
trunk setup for this Customer with our Service Provider (username below is
3035551122)
I don't see anywhere any config file the username = abc.com where could the
asterisk be picking it up from?
We have more than 10 such entries (all with same host = provider.sip.com value)
and when as INVITE is challenged, the Asterisk does match the correct trunk and
seems to send out correct Auth credentials...but not the one below..
[trunk_1]
;register to SP
allow = ulaw
;context = test
dialformat = ${EXTEN:1}
canreinvite = no
hasexten = no
hasiax = no
hassip = yes
host = provider.sip.com
insecure = very
port = 5060
registeriax = no
registersip = yes
trunkname = test
trunkstyle = customvoip
username = 3035551122
disallow = gsm,g726,alaw
contact = 3035551122
secret = x
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Initial Audio Cut off
What is the difference between this and the other option suggested below? Just put in: Answer() Wait(1.5) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle McKarns Sent: Friday, September 17, 2010 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Initial Audio Cut off Have you tried something like this exten x => 1,Answer() exten x => n,Wait(2) exten x=> n,(whatever you are doing now) Thanks, Lyle J. McKarns --- Network Engineering Team n|m Nexus Management 4 Industrial Parkway Suite 101 Brunswick, Maine 04011 Tel (USA) : 1 207 319 1105 Tel (UK) : 0207 100 4968 Fax: 1 207 725 8552 Nexus Management, Inc.│ Registered Office: 4 Industrial Parkway, Suite 101, Brunswick, Maine. 04011│Company No. 19891257D, Registered in Maine│ A member of the Nexus Management Plc group of companies -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo Sent: Friday, September 17, 2010 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Initial Audio Cut off We have already tried that...but still there is say 1.5 sec delay but the actual Audio first 2-4 secs still get cut off.. Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Friday, September 17, 2010 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Initial Audio Cut off Just put in: Answer() Wait(1.5) On Fri, Sep 17, 2010 at 11:00 AM, Ujjval Karihaloo wrote: > With some carriers the initial Audio (2-4 secs) seems to get cut off > when using a Auto Attendant or Conf Meetme. > > Is there any known remedies for that. Just want to know if others have > seen that esp. with Level 3. > > > > If Auto Attendant says - "Welcome to ABC bank" > > Caller only hears "Bank" > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Initial Audio Cut off
We have already tried that...but still there is say 1.5 sec delay but the actual Audio first 2-4 secs still get cut off.. Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Friday, September 17, 2010 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Initial Audio Cut off Just put in: Answer() Wait(1.5) On Fri, Sep 17, 2010 at 11:00 AM, Ujjval Karihaloo wrote: > With some carriers the initial Audio (2-4 secs) seems to get cut off when > using a Auto Attendant or Conf Meetme. > > Is there any known remedies for that. Just want to know if others have seen > that esp. with Level 3. > > > > If Auto Attendant says - "Welcome to ABC bank" > > Caller only hears "Bank" > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Initial Audio Cut off
Is DAHDI the Analog /PRI card..or something.. We never use it.. Call is delivered over SIP from the carrier...and plays the standard WAV file in Asterisk... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, September 17, 2010 9:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Initial Audio Cut off From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo Sent: Friday, September 17, 2010 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Initial Audio Cut off With some carriers the initial Audio (2-4 secs) seems to get cut off when using a Auto Attendant or Conf Meetme. Is there any known remedies for that. Just want to know if others have seen that esp. with Level 3. If Auto Attendant says - "Welcome to ABC bank" Caller only hears "Bank" Happens almost 100% of the time with a DAHDI connection (line supervision issue). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Initial Audio Cut off
With some carriers the initial Audio (2-4 secs) seems to get cut off when using a Auto Attendant or Conf Meetme. Is there any known remedies for that. Just want to know if others have seen that esp. with Level 3. If Auto Attendant says - "Welcome to ABC bank" Caller only hears "Bank" -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASterisk CDR file Master.csv
Thx Dean. I will be interested in testing that as well. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Hoover Sent: Friday, August 27, 2010 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ASterisk CDR file Master.csv On 8/27/2010 11:55 AM, Danny Nicholas wrote: > *From:* asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tim Nelson > *Subject:* Re: [asterisk-users] ASterisk CDR file Master.csv > > - "Ujjval Karihaloo" wrote: >> > >>How can we set the CDR Master file to rollover at say 30 Meg and create > a new one > > Use 'logrotate'. > > --Tim > > To "improve" on your answer, set up a shell to check the size of CDR > master and do a logrotate (/usr/sbin/asterisk -rx "logger rotate") when > the condition is met. GIYF on this one. > I use logrotate to help with the files in /var/log/asterisk, but it does nothing with Master.csv. I am working on a script described earlier that if the file gets larger than a certain number of lines to move them off to another file and compress for space. If that is something anyone else is interested in, let me know and I'll post it when it's working. -- Dean Hoover Milwaukee, Wisconsin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ASterisk CDR file Master.csv
How can we set the CDR Master file to rollover at say 30 Meg and create a new one -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call-limit field
If I set a call-limit field on a peer in users.conf.. I am seeing that it seems to affect other peers too? I am running Asterisk 1.4.18 has someone seen this issue. Peer 1 has call-limit=5 Peer 2 has call-limit=20... In the SIP trace I see when Peer2 hits 5, Asterisk sends back a 480 (temp Unavailable (Call limit reached)...msg.. Any ideas would be appreciated Thx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rolling over Master.csv CDR File
Is there a setting to roll over the Master.csv CDR File in /var/log/asterisk/cdr-csv, from and ZIP the older file once its gets a certain size? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] COnfig File question
1.7 for ASteriskNOw I will investigate..Thx for the ideas! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Thursday, August 05, 2010 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] COnfig File question On Thu, Aug 5, 2010 at 11:14 AM, Felipe Figueiredo mailto:felipe.figueired...@gmail.com>> wrote: Yes. Unless you use "make samples" while compiling the new Asterisk, you won't lose your confg files. I'm afraid there's no 1.7 version of Asterisk. [cid:image001.png@01CB34AA.02451E80] But there is a 1.7 version of AsteriskNow, which is what he was asking about. I'm not sure how you'd go about updating that though. -- Thanks, --Warren Selby http://www.selbytech.com <>-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] COnfig File question
Any answers would be appreciated Thx UK From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo Sent: Thursday, July 29, 2010 4:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] COnfig File question Hi All: If we upgrade asteriskNow from 1.4.18 to 1.7.0; just want to make sure all the config file functionality will reamin same That is - everything in /etc/asterisk will still work the same way. Users.conf Provider.conf Extensions.conf Sip.conf Etc... Thx in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] COnfig File question
Hi All: If we upgrade asteriskNow from 1.4.18 to 1.7.0; just want to make sure all the config file functionality will reamin same That is - everything in /etc/asterisk will still work the same way. Users.conf Provider.conf Extensions.conf Sip.conf Etc... Thx in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk unresponsive
Thx for all the responses. Really appreciate it. I will try putting the FQDN toIP address mapping in the /etc/hosts file to see if that makes a difference. I will also setup a cron to restart it every day... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: Wednesday, July 28, 2010 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk unresponsive Good call! I was just reading this thread and was preparing to write a reply mentioning DNS and SIP channel "lockups"... Basically, OP, Asterisk's SIP channels don't like not being able to do timely DNS queries, so you end up with a very unresponsive Asterisk server if you don't have local DNS caching [and][or] solid fast DNS servers available to you 24-7 I've been exactly where you are now, trust me, at first those "lockups" were causing me to lose sleep and think I was NUTS! Here's a helpful search: http://www.google.com/search?ie=UTF-8&oe=UTF-8&sourceid=navclient&gfns=1&q=asterisk-users+dns+sip+lockup Cheers, Sherwood McGowan ...I've been working with VoIP for almost 10 years now!?!?!AUUGH! On Wed, Jul 28, 2010 at 4:54 PM, Philipp von Klitzing wrote: >> We are running asteriskNow 1.4.18 and after a few days it becomes >> unresponsive and inbound INVITEs timeout. > > Search this list for "DNS". > > Philipp > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk unresponsive
We are running asteriskNow 1.4.18 and after a few days it becomes unresponsive and inbound INVITEs timeout. We just reboot the box to resolve it. But it seems to be occurring more regularly now. I am hesitant to move to latest version, but will do if needed. Any guidance or troubleshooting modes I may use will be helpful. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 negotiations in RTP
T38 rfc does not detail the sequence of t30 negotiations.. Like v21 preamble, nsf, Dsi, what the correct sequence of t30 negotiation. Is. On Oct 13, 2009, at 8:57 PM, "Kevin P. Fleming" wrote: > Ujjval Karihaloo wrote: >> Is there any t38 spec which details the T30 negotiations that occur >> in >> the Media/RTP PAth like the NSF, DSI etc frames that are exchanges >> between T38 gateways/endpoints. > > The T.38 recommendation defines how T.30 negotiations are carried in > IFP > (Internet Fax Protocol) packets; the definitions of the actual frames > are in the T.30 recommendation itself. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kpflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T38 negotiations in RTP
Is there any t38 spec which details the T30 negotiations that occur in the Media/RTP PAth like the NSF, DSI etc frames that are exchanges between T38 gateways/endpoints. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 free codec any idea
Dudes. just use G723... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Thursday, October 08, 2009 8:47 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] g729 free codec any idea On 9/10/09 3:31 PM, Michelle Dupuis wrote: > I believe that Intel placed a 729 codec into the public domain (free), > and someone wrapped it in a nice Asterisk package for use. > No idea where - but I do recall that it is out there, and legal. Of > course it's nice to support a vendor, but free alternatives can't be > shunned... The original comment stands. The codec is patented. The implementation is not. In order to use the implementation you need a license unless you live somewhere that: A) Doesn't have patents B) Doesn't have a trade agreement with USA Inserting a g729 codec from a licensed source other than Digium will break the GPL (Digium issues an exception for it's g729): http://tinyurl.com/ykpu42u This conversation has come up hundreds of times on this mailing list and the result is always the same - if you're happy breaking the law, go for it - if you get most of your movies from piratebay then it probably isn't a problem for you. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 REINVITe issue
Already have it...
If provider does not challenge re- invite
Fax works fine!
Ujjval
On Oct 6, 2009, at 11:33 PM, "Trevor Peirce"
wrote:
> Ujjval Karihaloo wrote:
>>
>> Her eis my users.conf entry for Asterisk registration to the Sip
>> Provider. (I know I don’t have T38 as allowed codecs, not sure wha
>> t to
>> add for T38)
>>
>> [trunk_66]
>>
>> ;register
>>
>> allow = ulaw
>>
>> dialformat = ${EXTEN:1}
>>
>> canreinvite = no
>>
>> hasexten = no
>>
>> hasiax = no
>>
>> hassip = yes
>>
>> host = provider.com
>>
>> insecure = very
>>
>> port = 5060
>>
>> registeriax = no
>>
>> registersip = yes
>>
>> trunkname = abc
>>
>> username = abc
>>
>> disallow = gsm,g726,alaw
>>
>> contact = abc
>>
>> secret = abc
>>
>
> You'll need to put t38pt_udptl = yes somewhere in your sip.conf,
> probably in the general section for T.38 to work properly.
>
> --
> Trevor Peirce
> Digital Conceptions Canada
>
> http://www.digitalcon.ca
> 1-888-606-3030 / 250-391-7822
>
>
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Re: [asterisk-users] T38 REINVITe issue
Anyone for this ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo
Sent: Monday, October 05, 2009 11:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] T38 REINVITe issue
Hi
My call flow is
T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN
Call is placed in reverse direction - from PSTN to T38 Gateway.
T38 GW reinvites with T38, and asterisk passes it along to the SIP provider.
The SIP provider challenges it and asterisk reponds to the Challenge with
INVITE with Auth credentials...however, the Asterisk changes the SDP and
replaces the T38 info in SDP with G711uLawand fax fails. How do I configure
the host entry in users.conf such that it maintains the T38 reinvite as it
responds to the SIP INVITE challenge from the Sip Provider.
Her eis my users.conf entry for Asterisk registration to the Sip Provider. (I
know I don't have T38 as allowed codecs, not sure what to add for T38)
[trunk_66]
;register
allow = ulaw
dialformat = ${EXTEN:1}
canreinvite = no
hasexten = no
hasiax = no
hassip = yes
host = provider.com
insecure = very
port = 5060
registeriax = no
registersip = yes
trunkname = abc
username = abc
disallow = gsm,g726,alaw
contact = abc
secret = abc
Any ideas appreciated.
Thx
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[asterisk-users] T38 REINVITe issue
Hi
My call flow is
T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN
Call is placed in reverse direction - from PSTN to T38 Gateway.
T38 GW reinvites with T38, and asterisk passes it along to the SIP provider.
The SIP provider challenges it and asterisk reponds to the Challenge with
INVITE with Auth credentials...however, the Asterisk changes the SDP and
replaces the T38 info in SDP with G711uLawand fax fails. How do I configure
the host entry in users.conf such that it maintains the T38 reinvite as it
responds to the SIP INVITE challenge from the Sip Provider.
Her eis my users.conf entry for Asterisk registration to the Sip Provider. (I
know I don't have T38 as allowed codecs, not sure what to add for T38)
[trunk_66]
;register
allow = ulaw
dialformat = ${EXTEN:1}
canreinvite = no
hasexten = no
hasiax = no
hassip = yes
host = provider.com
insecure = very
port = 5060
registeriax = no
registersip = yes
trunkname = abc
username = abc
disallow = gsm,g726,alaw
contact = abc
secret = abc
Any ideas appreciated.
Thx
___
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Register Now: http://www.astricon.net
asterisk-users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
