[Asterisk-Users] Latest CVS, Grandstream and Zaptel bug?

Peter Boot peter.boot at webfone.com.au
Thu Jun 24 22:38:01 MST 2004


I had the same problem when using a Grandstream 486 I solved it by using the
nat=yes config option

>>-----Original Message-----
>>From: asterisk-users-admin at lists.digium.com 
>>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
>>Andrew Yager
>>Sent: Friday, June 25, 2004 3:31 PM
>>To: asterisk-users at lists.digium.com
>>Subject: [Asterisk-Users] Latest CVS, Grandstream and Zaptel bug?
>>
>>Hi,
>>
>>I'm confused as anything by this bug. I'm hoping that it is 
>>just something screwy in my config.
>>
>>I have 1 Cisco 7960 and several Grandstream BT101 & 102's, 
>>and a Digium TDM31B.
>>
>>I'm running the latest CVS (CVS-HEAD-05/27/04-17:22:40) of 
>>both Asterisk and the Zaptel driver on Fedora Cora 1.
>>
>>When I make an outgoing call on the Cisco phone, everything 
>>works fine. 
>>I'm connected, and it all sounds hunky dory.
>>
>>My Grandstream phones talk quite nicely to Asterisk. I can 
>>receive incoming calls and have them forwarded to my phone, 
>>and I can dial internal extensions without a problem. 
>>However, whenever I attempt to make an outgoing call, the 
>>outgoing number rings, but no audio is ever sent to the 
>>Grandstreams, even when the phone is answered. If I put an r 
>>in the dial plan, the GrandStream does generate the ringing tone. 
>>When an m is set, no audio is transmitted to the phone. The 
>>person who answers the call hears absolutely nothing at all.
>>
>>The Grandstream phones can talk to each other without a problem.
>>
>>It seems that the bug is being generated between the 
>>Grandstream phones and the Zap card, but only on outgoing calls.
>>
>>To add to the confusion, if I phone one of the FXS ports 
>>connected to our hard fax, it rings, answers and everything 
>>works just fine.
>>
>>My zapata.conf is presently:
>>
>>[channels]
>>context=incoming
>>signalling=fxs_ls
>>rxwink=300              ; Atlas seems to use long (250ms) winks
>>usecallerid=no
>>hidecallerid=no
>>callwaiting=no
>>usecallingpres=yes
>>callwaitingcallerid=no
>>threewaycalling=yes
>>transfer=yes
>>cancallforward=yes
>>callreturn=yes
>>echocancel=yes
>>echocancelwhenbridged=no
>>echotraining=yes ; have tried changing this to yes and 800 - 
>>no difference rxgain=0.0 txgain=0.0
>>group=1
>>callgroup=1
>>pickupgroup=1
>>immediate=no
>>busydetect=no ; previously had this at yes, but when changed 
>>it to no to test
>>busycount=4
>>musiconhold=default
>>faxdetect=incoming
>>channel => 1
>>
>>Any help or suggestions on what to try or where to go would 
>>be appreciated.
>>
>>Andrew
>>
>>_________________________
>>Andrew Yager
>>Real World Technology Solutions
>>Real People, Real SolUtions (tm)
>>ph: (02) 9945 2567 fax: (02) 9945 2566
>>mob: 0405 15 2568
>>http://www.rwts.com.au/
>>_________________________
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at 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
>>
>>---
>>Incoming mail is certified Virus Free.
>>Checked by AVG anti-virus system (http://www.grisoft.com).
>>Version: 6.0.710 / Virus Database: 466 - Release Date: 6/23/2004
>> 
>>

---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.710 / Virus Database: 466 - Release Date: 6/23/2004
 




More information about the asterisk-users mailing list