[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 -- _ -- Band

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 th

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

[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.co

Re: [asterisk-users] Initial Audio Cut off

e.  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

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

Re: [asterisk-users] Initial Audio Cut off

, 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

[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" -- __

Re: [asterisk-users] ASterisk CDR file Master.csv

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

[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:

[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

[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 A

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:

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

[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. --

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:a

[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 mod

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 nego

[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 -- AstriCo

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 cod

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 k

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

[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 w