Re: [Asterisk-Users] SIP to IAX - forcing codec pass thru
Mark Phillips wrote: Hi Peter, I don't see any codec allow=blah statements. If your end user has something like [gradwell] disallow=all allow=gsm Then you'll be forced to send them a GSM coded call. Why not force the codec at your end by only supporting one? If the customer then transcodes the call when it gets forwarded to his handset there's not much you can do about that but at least you'll have handed the call off in the best format you can source. mmm, but as you've seen, some customers like using multiple codecs. The cisco kit is able to support a raft of options - and it does transcoding very nicely - so the optimum solution is to have the cisco + customer's asterisk agree on the same codec, and then have our asterisk server (in the middle) do as little as possible. cheers peter -- peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/ -- engineering & hosting services for email, web and voip -- -- http://www.peter.me.uk/ -- http://www.voip.org.uk/ -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP to IAX - forcing codec pass thru
hi
We take calls inbound via SIP from our Cisco PSTN gateways, and pass it
to customers using IAX (they run their own asterisk servers).
We've noticed that asterisk is transcoding the call into a different
codec, if the customer prefers a codec different to that which our cisco
gw prefers. As such, the quality of the call can degrade.
We'd rather asterisk just passed through the RTP stream and maintained
the same codec, so that all asterisk did was signalling conversion.
sip.conf...
---
[sip-router-1.gradwell.net]
context=sip-inbound
type=peer
host=sip-router-1.gradwell.net
[sip-router-2.gradwell.net]
context=sip-inbound
type=peer
host=sip-router-2.gradwell.net
---
iax.conf...
[general]
bandwidth=high
disallow=lpc10
jitterbuffer=yes
dropcount=2
maxjitterbuffer=500
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1
tos=lowdelay
---
when a call comes in, we dial something like this, in our dial plan:
-- Executing Goto("SIP/213.166.5.134-118f5310",
"sip-users|7770002|1") in new stack
-- Goto (sip-users,7770002,1)
-- Executing Dial("SIP/213.166.5.134-118f5310",
"IAX2/user:[EMAIL PROTECTED]/441376350002") in new stack
-- Called user:[EMAIL PROTECTED]/441376350002
-- Call accepted by customerip (format alaw)
-- Format for call is alaw
-- IAX2/customerip:4569-23 answered SIP/213.166.5.134-118f5310
thanks
peter
--
peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/
-- engineering & hosting services for email, web and voip --
-- http://www.peter.me.uk/ -- http://www.voip.org.uk/ --
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[Asterisk-Users] SER/Asterisk PSTN Call Transfer Issue.
Hi We have a phone system consisting primarily of SER and Asterisk, and are having trouble transferring inbound calls from the PSTN. We believe the problem is basically that because our phones register with SER, the Asterisk box never sees the call from the original callee to the new callee. i.e. caller --> PSTN --> Asterisk --> SER --> callee's SIP phone callee's SIP phone --> SER --> new callee's SIP phone when the original caller hangs up, the transfer is requested, however Asterisk does not know anything about the second call shown above, and so we see an error about "supervised transfer requested; callid not found". We have found that if, in our SER config, we route 8 through Asterisk, having stripped off the leading '8', and on to the SIP phone via SER, then it works, because Asterisk then 'knows' about the new call, since the second call above becomes callee's SIP phone --> SER --> Asterisk --> SER --> new callee's SIP phone Of course, this is a rather inelegant fix, especially when we actually have more than one PSTN gateway. Does anyone know if there is a more flexible fix, either on the SER or Asterisk side, that might fix our problem? thanks peter -- peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/ -- engineering & hosting services for email, web and voip -- -- http://www.peter.me.uk/ -- http://www.voip.org.uk/ -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
