[Serusers] Re: [cisco-voip] no inward calls from PSTN possible
Gerd Feiner
g.feiner at cablesurf.de
Sat May 7 18:22:39 CEST 2005
Additional info:
when the disconnect from the isdn occurs the following appears in the
log:
May 7 18:12:42 195.234.128.23 65018: ^ICause i = 0x82BF -
Service/option not available, unspecified
Brgds,
Gerd
Am 07.05.2005 um 15:59 schrieb Gerd Feiner:
> Hi,
>
> (sorry for posting this to two different lists, but I think there
> maybe different considerations of this problem)
>
> I have an SER setup with mediaproxy and a Cisco AS5350. The
> SER/mediaproxy is setup following the guide at www.onsip.org. The
> AS5350 does just simple mediaconversion (so there are two dial-peers,
> one for PSTN and one for VOIP). These dialpeers worked fine with a
> less tightened SER-config, but always had the problem of not giving me
> any dialtone.
>
> Now, when I dial inwards from PSTN there is a very short dialtone
> which gets interrupted by silence. So far so good. But, when I pick
> up the SIP-phone I get an error-message on my PSTN-phone telling that
> the service was unavailable (yes - I payed my bill :--) and the
> SIP-phone gets disconnected.
>
> When debugging the AS5350 i noticed that is does net get an rport when
> dialing inwards:
>
> voip_rtp_create_session: callID=277, dstCallID=-1
> laddr=195.234.128.23, lport=18114,raddr=0.0.0.0, rport=0, type=2,
> sig_tos=3, ip_tos=5
>
> When the SIP-phone is picked up, the AS5350 gets a new information
> regarding the rport:
>
> May 7 15:39:28 195.234.128.23 57384: ^I State :
> STREAM_ADDING (2)
> May 7 15:39:28 195.234.128.23 57385: ^I Callid : 277
> May 7 15:39:28 195.234.128.23 57386: ^I Negotiated Codec :
> g711alaw, bytes :160
> May 7 15:39:28 195.234.128.23 57387: ^I Negotiated DTMF relay :
> inband-voice
> May 7 15:39:28 195.234.128.23 57388: ^I Negotiated NTE payload : 0
> May 7 15:39:28 195.234.128.23 57389: ^I Negotiated CN payload : 0
> May 7 15:39:28 195.234.128.23 57390: ^I Media Srce Addr/Port :
> 195.234.128.23:18114
> May 7 15:39:28 195.234.128.23 57391: ^I Media Dest Addr/Port :
> 62.141.42.76:16032
>
> but immediately after that the call gets disconnected.
>
> When dialing from SIP to PSTN everything seems fine with the rport:
>
> voip_rtp_create_session: callID=278, dstCallID=279
> laddr=195.234.128.23, lport=17496,raddr=62.141.42.76, rport=16034,
> type=3, sig_tos=3, ip_tos=5
>
> I also noticed that there is a difference in the type-value: 2 when
> dialing inwards and 3 when dialing outwards.
>
> Any ideas why this is happening?
>
> Brgds,
> Gerd
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
A non-text attachment was scrubbed...
Name: PGP.sig
Type: application/pgp-signature
Size: 186 bytes
Desc: Signierter Teil der Nachricht
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20050507/ea992ae2/attachment.pgp>
More information about the sr-users
mailing list