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...ation "unknown" set 6 ntp server 192.168.30.1 set 6 ntp timezone 1 set 6 ntp enable set 6 cdr enable set 6 hookflash 0 set 6 mgcp addressformat nobrackets set 6 mgcp callagent address 192.168.30.1 set 6 mgcp callagent port 2427 set 6 mgcp up set 6 mgcp rsipwildcard enable set 6 voip ptime g711mu 10 set 6 voip ptime g711a 10 set 6 voip rtcp cname "adit" set 6 compander alaw set 6 voip sdpaddress gatewayid set 6:1:1:1 log start mgcp set 6:1:1:1-48 algorithm preference g711mu g711a g726_16 g726_24 g726_32 \ g726_40 set 6:1:1:4 echo cancellation disable set 6:1:1:4 fax bypass set 6:1...

Hi! There's any way to set up a call using G726 (sip peer) receive it on Asterisk convert it to G711Mu to send it to PSTN broadband termination? I've put the following in sip.conf: disalow=all allow=gsm allow=g726 (my TAs use G726 32K) best regards, Helder -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attac...

...7 set 6 mgcp gatewayid 10.0.0.245 set 6 mgcp quarantine step discard set 6 mgcp port 2727 set 6 mgcp up set 6 mgcp rsipwildcard enable set 6 mgcp tos 0x68 set 6 voip osi 500 set 6:1:1:1 log start both set 6:1:1:1-48 echo tail 64 set 6:1:1:1-48 tos 0xB8 set 6:1:1:1-48 algorithm preference g711mu g729a set 6:1:1:1-48 dtmfrelay enable set 6:1:1:1-48 cpd osi reset 6 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090630/6dc5fcb4/attachment.htm

Anyone using the G729 codec to create a h.323 trunk between an Avaya Communication manager and Asterisk Freepbx System and works? I don't have the G729 codec installed on the Asterisk and running G711MU on avaya and getting invalid codec when calling from Avaya to Asterisk. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120130/a90a704d/attachment.htm>

I have a Cisco 2600 which understands the "gsmfr" codec, which appears to be what Asterisk calls "gsm" -- at least it ends up using it. I also have a PSTN gateway which is speaking ulaw. When the 2600 calls through Asterisk to the PSTN, it negotiates the g711ulaw codec, but when the PSTN calls through Asterisk to the 2600, it seems that Asterisk is doing translation, and it

Hi all - This isn't really directly Asterisk related, but has anyone successfully set up a Polycom phone to register two lines on two different Asterisk boxes? I can get the first line to register, but the second one does not. I can still place calls from that second line, which indicates to me the server, user, and secret are correct. I'm running the newest 2.6 series firmware with the

...default Capability: 0xf (g723|gsm|ulaw|alaw) DTMF Mode: rfc2833 AccountCode: ast_h323 AMA flags: Unknown Aliases: 100 ObjSysAsterisk Avaya codecs setting: ------------------------------- 1 G729 2 g711a 3 g711mu /etc/asterisk/ooh323.conf ----------------------- [avaya] type=peer context=default ip=192.168.0.14 ; UPDATE with appropriate ip address port=1720 ; UPDATE with appropriate port e164=101 disallow=all allow=ulaw allow=alaw /var/log/asterisk/h323_log ----------------------------- 18:44:37:802...