[Asterisk-Users] SEND TEXT to an extension?
Hi, I understand SendText() sends text on the current channel. Is there a way to manipulate this feature to SendText toward another SIP device? I use Polycom IP600's. Local sendtext works fine. Would be nice to drop an instant message on another user's phone. thanks! Mark ___ 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] CODEC Allow statement help
Hello, I have 6 Asterisk switches all running together nicely with DUNDi and have one minor problem with inter switch CODEC negotiation. I use G729 (licensed from Digium) on several of the switches. Inbetween the G729 switches we can make calls no problem. >From a switch that only does ULAW they cannot make a call into my G729 switch. (the call fails with an RTP translation error) The G729 switch can build a call toward the ULAW switch, and the call is processed as ULAW. On my G729 switches: both sip/iax.conf show disallow=all allow=g729 allow=ulaw On the ULAW switches: disallow=all allow=ulaw I have tried every odd combination of allow statements for the ULAW switches to build a call while still allowing the G729 switches to build calls. The phones are a mixture of Budgettones and Polycom IP600's. The phones all have thier first codec set to g729. the second to ULAW. My question is this, in a multi CODEC environment where some phones are ULAW, some are G729, and some switches are ULAW, some are G729 licensed, what is the best set of statements to get them all to play together? thanks! Mark ___ 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] Polycom IP600 Cannot answer - SOLVED
SOLVED! By updating my CVS head just now, my Polycom IP 600 works great! Thank you! Mark MDS wrote: >> I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6 >> months, no problems with my grandstreams. I'm fairly familiar with the >> ins and outs of asterisk... > > From: "Kevin P. Fleming If you are going to use CVS HEAD, you _must_ stay up to date. There have been a large number of SIP-related fixes in CVS HEAD since the 19th. Please update your system and try again before reporting problems. Thanks! ___ 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] Polycom IP600 Cannot answer
I googled and googled but could not find anything regarding this problem.
I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6
months, no problems with my grandstreams. I'm fairly familiar with the
ins and outs of asterisk...
IP600 with latest sip 1.4.1 and bootrom from my FTP server.
Standard config files from http://www.freedomphones.net/polycom/files/
No changes other than typical ip address of phone and server.
Grandstream (192.168.2.20) is exten 2000, Polycom (192.168.2.22) is 2006.
I can make calls out to my Grandstreams from the Polycom all day. No
problem.
When I try to call the Polycom I get this stuff:
-- Executing Dial("SIP/2000-972f", "SIP/2006|10|r") in new stack
-- Called 2006
-- SIP/2006-f8ea is ringing
-- SIP/2006-f8ea answered SIP/2000-972f
-- Attempting native bridge of SIP/2000-972f and SIP/2006-f8ea
-- Got SIP response 481 "No Such Call" back from 192.168.2.20
== Spawn extension (from-sip, 2006, 1) exited non-zero on 'SIP/2000-972f'
-- Got SIP response 500 "Internal Server Error" back from 192.168.2.22
-- Got SIP response 500 "Internal Server Error" back from 192.168.2.22
When I answer the polycom it just hangs up and hangs the grandstream
online. I have to manually hang up the grandstream. It doesn't get a SIP
notifcation of call failure or hangup.
When I tcpdump the asterisk box, I can see RTP streams from the
Grandstream toward the server. But nothing coming from or toward the
Polycom. When I call the Grandstream from the Polycom, the call connects
and I see both RTP streams to and from the Asterisk box for both phones
and everything is happy.
anyone have any ideas as to why inbound calls fail?
I've tried several combinations of
friend/peer/progressinband/canreinvite etc... No change at all.
Here's my sip.conf for the Polycom
[2006]
type=friend
username=2006
secret=2006
host=dynamic
dtmfmode=rfc2833
defaultip=192.168.2.22
progressinband=no
context=from-sip
[EMAIL PROTECTED]
callgroup=1
pickupgroup=1
thank you for any insight!
Mark
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