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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" -------------- next part -------------- An HTML attachment was scrubbed... URL:

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. -------------- next part -------------- An HTML attachment was

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

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

How can we set the CDR Master file to rollover at say 30 Meg and create a new one -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/2e98385f/attachment.htm

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 -------------- next

Hi Below is my register string register => username at test.abc.com:xxxxxxx:uername at test.server.com The REGISTER from asterisk has the From header with test.server.com instead of test.abc.com Any help appreciated. UK -------------- next part -------------- An HTML attachment was scrubbed... URL: