Bryan M. Johns
2005-Jun-16 19:39 UTC
[Asterisk-Users] Routing SIP to Cisco routers running IOS 12.3+
I've experiencing some difficulty passing inbound calls from the PSTN, through a large Asterisk switch and down our network to a Cisco 1751 router. This router has 4 FXS ports and is running IOS 12.3. Outbound dialing from phones on the FXS ports of the router works flawlessly, but inbound calls fail as though the Asterisk server does not see the extensions representing the FXS ports as available or registered. There is little to lead me to believe that IOS will support a port-over-port SIP registration with Asterisk, so I have configured sip_additional.conf with the following format for each extension on the 1751: [XXXXXXXXXX] username=XXXXXXXXXX type=friend port=5060 nat=yes host=xxx.xxx.xxx.xxx dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="XXXXXXXXXXX" <XXXXXXXXXXX> allow=alaw I am relatively confident that my problem does not exist at the 1751 due to the ability to flawlessly process outgoing calls. However, after more than a day in this one, I guess anything is possible. Does anybody out there have any experiencing sending SIP down-wire from Asterisk to the Cisco IOS? We might be willing to pay for the right kind of help here. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050616/dc66ca1b/attachment.htm
Hi all, After upgrading to latest CVS head, I have problems using a IAXY device, having slin problems: Jun 15 18:59:31 NOTICE[8197]: channel.c:1475 ast_read: Dropping incompatible voice frame on IAX2/lise-1 of format slin since our native format has changed to ulaw Because of that outside caller can't ear the callee on the IAXY. Found somewhere that disabling transcode in asterisk.conf would fix the problem, so I added, not sure of the syntax the following section in asterisk.conf: [options] transcode_via_sln=no That didn't work, and I am not sure I am using the wright syntax... I have revert back to stable release and everything is ok, but I want to test SCOPSERV-VoIP and it requires version 1.07 or higher... Regards, Francois Random Thought: --------------- The supreme accomplishment is to blur the line between work and play. - Arnold Joseph Toynbee, 1889 - 1975