Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code
On Oct 9, 2008, at 10:40 AM, [EMAIL PROTECTED]
wrote:
> Hi I have searched the mailing lists and come across similar threads,
> but no actual solution. I am trying to use a Cisco AS5300 as a
> gateway for PSTNr. I have been able to configure it to take outbound
> calls and send them to the PSTN just fine. Inbound calls however are
> rejected by asterisk with "488 Not acceptable here" code.
>
> here are the details:
>
> AS5300:
> IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE
> SOFTWARE (fc5)
Make sure the context defined the for the cisco exists and matches the
extensions. Try a catch all pattern: exten => _.*,1,Nop(${EXTEN})
I had some random issues and I ended up defining another entry in
SIP.conf for the cisco by by IP address, e.g.
[172.31.2.7]
type=friend
host=172.31.2.7
insecure=very
context=cisco
qualify=2000
dtmfmode=inband
That works for me.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
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Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code
On 10/9/08, Ketema Harris <[EMAIL PROTECTED]> wrote: > > Hi I have searched the mailing lists and come across similar threads, but no > actual solution. I am trying to use a Cisco AS5300 as a gateway for PSTNr. > I have been able to configure it to take outbound calls and send them to the > PSTN just fine. Inbound calls however are rejected by asterisk with "488 > Not acceptable here" code. > > here are the details: > ...lots of details stripped... Thanks for the all of the info. Wading through the various debugs I would guess that Asterisk 1.2's SDP parser does not like the multipart INVITE from the AS5300 with MIME type multipart/mixed, followed by a regular SDP (application/sdp) and GTD (application/gtd) (ew). You have two choices: 1) Disable GTD on the AS5300. I personally don't like GTD and I doubt you need the extra ISUP info anyway. I don't see GTD specifically enabled on the AS5300 but I only have experience with AS5X50s on more recent versions of IOS so there might be a default I don't know about... 2) Update/upgrade to Asterisk 1.4: http://bugs.digium.com/view.php?id=10947 Asterisk 1.4 has much better support for multipart/mixed. While this bug references SIP-T, the underlying problem (multipart messages) is the same. Let us know what happens. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code
Not offhand / without seeing the Asterisk side. On Thu, October 9, 2008 10:26 am, Ketema Harris wrote: > dtmf mode was set in the sip.conf > > dtmfmode=rfc2833 > > I will remove the other codecs from sip.conf and see what effect it > has. Do you see any other potential issues in the configs? > > thanks > > > On Oct 9, 2008, at 9:36 AM, Alex Balashov wrote: > >> >> This is due to an SDP mismatch of some sort, codec or otherwise. >> >> Perhaps you have not set your Asterisk SIP peers to support RFC2833 >> DTMF? Try dtmfmode=rfc2833. Either that, or your Asterisk SIP peers >> are not accepting the gateway's offer of G.711u. >> >> Of course, I have seen interop bugs in Asterisk in the past where >> inbound >> calls from Cisco ISDN gateways whose SDP payload advertises a >> different >> preferred codec--but one that is still acceptable further down the >> preference chain--is denied. You may want to try to set both sides to >> G.711u explicitly, i.e. >> >> disallow=all >> allow=ulaw >> >> On the Asterisk side. Also make sure dtmfmode is set. >> >> On Thu, October 9, 2008 9:25 am, Ketema Harris wrote: >> >>> Hi I have searched the mailing lists and come across similar threads, >>> but no actual solution. I am trying to use a Cisco AS5300 as a >>> gateway for PSTNr. I have been able to configure it to take outbound >>> calls and send them to the PSTN just fine. Inbound calls however are >>> rejected by asterisk with "488 Not acceptable here" code. >>> >>> here are the details: >>> >>> AS5300: >>> IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE >>> SOFTWARE (fc5) >>> >>> Current configuration : 3939 bytes >>> >>> version 12.3 >>> service timestamps debug datetime msec >>> service timestamps log datetime msec >>> no service password-encryption >>> ! >>> hostname K_AS5300_3 >>> ! >>> boot-start-marker >>> boot-end-marker >>> ! >>> enable password ** >>> ! >>> resource-pool disable >>> clock timezone EST -5 >>> clock summer-time EDT recurring >>> ! >>> no aaa new-model >>> ip subnet-zero >>> ! >>> ! >>> isdn switch-type primary-dms100 >>> ! >>> ! >>> voice service voip >>> sip >>> bind all source-interface FastEthernet0 >>> >>> controller T1 0 >>> framing esf >>> clock source line primary >>> linecode b8zs >>> pri-group timeslots 1-24 >>> ! >>> controller T1 1 >>> framing esf >>> clock source line secondary 1 >>> linecode b8zs >>> pri-group timeslots 1-24 >>> ! >>> controller T1 2 >>> framing esf >>> linecode b8zs >>> pri-group timeslots 1-24 >>> ! >>> controller T1 3 >>> framing esf >>> linecode b8zs >>> pri-group timeslots 1-24 >>> ! >>> ! >>> ! >>> interface Ethernet0 >>> no ip address >>> shutdown >>> ! >>> interface Serial0:23 >>> no ip address >>> encapsulation hdlc >>> isdn switch-type primary-dms100 >>> isdn incoming-voice modem 64 >>> no cdp enable >>> ! >>> interface Serial1:23 >>> no ip address >>> encapsulation hdlc >>> isdn switch-type primary-dms100 >>> isdn incoming-voice modem 64 >>> no cdp enable >>> ! >>> interface Serial2:23 >>> no ip address >>> encapsulation hdlc >>> isdn switch-type primary-dms100 >>> isdn incoming-voice modem 64 >>> no cdp enable >>> ! >>> interface Serial3:23 >>> no ip address >>> encapsulation hdlc >>> isdn switch-type primary-dms100 >>> isdn incoming-voice modem 64 >>> no cdp enable >>> ! >>> interface FastEthernet0 >>> ip address 172.31.2.7 255.255.255.0 >>> duplex auto >>> speed auto >>> ! >>> ip classless >>> ip route 0.0.0.0 0.0.0.0 172.31.2.1 >>> no ip http server >>> ! >>> ! >>> ! >>> ! >>> ! >>> ! >>> voice-port 0:D >>> ! >>> voice-port 1:D >>> ! >>> voice-port 2:D >>> ! >>> voice-port 3:D >>> ! >>> ! >>> ! >>> dial-peer voice 100 voip >>> application session >>> destination-pattern 678... >>> signaling forward unconditional >>> session protocol sipv2 >>> session target sip-server >>> session transport udp >>> dtmf-relay rtp-nte >>> codec g711ulaw >>> no vad >>> ! >>> dial-peer voice 101 voip >>> destination-pattern 770... >>> progress_ind setup enable 3 >>> session protocol sipv2 >>> session target sip-server >>> session transport udp >>> dtmf-relay rtp-nte >>> codec g711ulaw >>> no vad >>> ! >>> dial-peer voice 102 voip >>> destination-pattern 404... >>> progress_ind setup enable 3 >>> session protocol sipv2 >>> session target sip-server >>> session transport udp >>> dtmf-relay rtp-nte >>> codec g711ulaw >>> no vad >>> ! >>> dial-peer voice 103 voip >>> destination-pattern 470... >>> progress_ind setup enable 3 >>> session protocol sipv2 >>> session target sip-server >>> session transport udp >>> dtmf-relay rtp-nte >>> codec g711ulaw >>> no vad >>> ! >>> dial-peer voice 200 pots >>> application session >>> incoming called-number . >>> destination-pattern 91.. >>> direct-inward-dial >>> port 0:D >>> prefix 1 >>> ! >>> dial-peer voice 201 pots >>> application session >>> incoming called-number . >>> destination-pattern 9.. >
Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code
dtmf mode was set in the sip.conf dtmfmode=rfc2833 I will remove the other codecs from sip.conf and see what effect it has. Do you see any other potential issues in the configs? thanks On Oct 9, 2008, at 9:36 AM, Alex Balashov wrote: > > This is due to an SDP mismatch of some sort, codec or otherwise. > > Perhaps you have not set your Asterisk SIP peers to support RFC2833 > DTMF? Try dtmfmode=rfc2833. Either that, or your Asterisk SIP peers > are not accepting the gateway's offer of G.711u. > > Of course, I have seen interop bugs in Asterisk in the past where > inbound > calls from Cisco ISDN gateways whose SDP payload advertises a > different > preferred codec--but one that is still acceptable further down the > preference chain--is denied. You may want to try to set both sides to > G.711u explicitly, i.e. > > disallow=all > allow=ulaw > > On the Asterisk side. Also make sure dtmfmode is set. > > On Thu, October 9, 2008 9:25 am, Ketema Harris wrote: > >> Hi I have searched the mailing lists and come across similar threads, >> but no actual solution. I am trying to use a Cisco AS5300 as a >> gateway for PSTNr. I have been able to configure it to take outbound >> calls and send them to the PSTN just fine. Inbound calls however are >> rejected by asterisk with "488 Not acceptable here" code. >> >> here are the details: >> >> AS5300: >> IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE >> SOFTWARE (fc5) >> >> Current configuration : 3939 bytes >> >> version 12.3 >> service timestamps debug datetime msec >> service timestamps log datetime msec >> no service password-encryption >> ! >> hostname K_AS5300_3 >> ! >> boot-start-marker >> boot-end-marker >> ! >> enable password ** >> ! >> resource-pool disable >> clock timezone EST -5 >> clock summer-time EDT recurring >> ! >> no aaa new-model >> ip subnet-zero >> ! >> ! >> isdn switch-type primary-dms100 >> ! >> ! >> voice service voip >> sip >> bind all source-interface FastEthernet0 >> >> controller T1 0 >> framing esf >> clock source line primary >> linecode b8zs >> pri-group timeslots 1-24 >> ! >> controller T1 1 >> framing esf >> clock source line secondary 1 >> linecode b8zs >> pri-group timeslots 1-24 >> ! >> controller T1 2 >> framing esf >> linecode b8zs >> pri-group timeslots 1-24 >> ! >> controller T1 3 >> framing esf >> linecode b8zs >> pri-group timeslots 1-24 >> ! >> ! >> ! >> interface Ethernet0 >> no ip address >> shutdown >> ! >> interface Serial0:23 >> no ip address >> encapsulation hdlc >> isdn switch-type primary-dms100 >> isdn incoming-voice modem 64 >> no cdp enable >> ! >> interface Serial1:23 >> no ip address >> encapsulation hdlc >> isdn switch-type primary-dms100 >> isdn incoming-voice modem 64 >> no cdp enable >> ! >> interface Serial2:23 >> no ip address >> encapsulation hdlc >> isdn switch-type primary-dms100 >> isdn incoming-voice modem 64 >> no cdp enable >> ! >> interface Serial3:23 >> no ip address >> encapsulation hdlc >> isdn switch-type primary-dms100 >> isdn incoming-voice modem 64 >> no cdp enable >> ! >> interface FastEthernet0 >> ip address 172.31.2.7 255.255.255.0 >> duplex auto >> speed auto >> ! >> ip classless >> ip route 0.0.0.0 0.0.0.0 172.31.2.1 >> no ip http server >> ! >> ! >> ! >> ! >> ! >> ! >> voice-port 0:D >> ! >> voice-port 1:D >> ! >> voice-port 2:D >> ! >> voice-port 3:D >> ! >> ! >> ! >> dial-peer voice 100 voip >> application session >> destination-pattern 678... >> signaling forward unconditional >> session protocol sipv2 >> session target sip-server >> session transport udp >> dtmf-relay rtp-nte >> codec g711ulaw >> no vad >> ! >> dial-peer voice 101 voip >> destination-pattern 770... >> progress_ind setup enable 3 >> session protocol sipv2 >> session target sip-server >> session transport udp >> dtmf-relay rtp-nte >> codec g711ulaw >> no vad >> ! >> dial-peer voice 102 voip >> destination-pattern 404... >> progress_ind setup enable 3 >> session protocol sipv2 >> session target sip-server >> session transport udp >> dtmf-relay rtp-nte >> codec g711ulaw >> no vad >> ! >> dial-peer voice 103 voip >> destination-pattern 470... >> progress_ind setup enable 3 >> session protocol sipv2 >> session target sip-server >> session transport udp >> dtmf-relay rtp-nte >> codec g711ulaw >> no vad >> ! >> dial-peer voice 200 pots >> application session >> incoming called-number . >> destination-pattern 91.. >> direct-inward-dial >> port 0:D >> prefix 1 >> ! >> dial-peer voice 201 pots >> application session >> incoming called-number . >> destination-pattern 9.. >> direct-inward-dial >> port 0:D >> ! >> dial-peer voice 202 pots >> application session >> incoming called-number . >> destination-pattern 91.. >> direct-inward-dial >> port 1:D >> prefix 1 >> ! >> dial-peer voice 203 pots >> application session >> incoming called-number . >> destination-patt
Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code
This is due to an SDP mismatch of some sort, codec or otherwise. Perhaps you have not set your Asterisk SIP peers to support RFC2833 DTMF? Try dtmfmode=rfc2833. Either that, or your Asterisk SIP peers are not accepting the gateway's offer of G.711u. Of course, I have seen interop bugs in Asterisk in the past where inbound calls from Cisco ISDN gateways whose SDP payload advertises a different preferred codec--but one that is still acceptable further down the preference chain--is denied. You may want to try to set both sides to G.711u explicitly, i.e. disallow=all allow=ulaw On the Asterisk side. Also make sure dtmfmode is set. On Thu, October 9, 2008 9:25 am, Ketema Harris wrote: > Hi I have searched the mailing lists and come across similar threads, > but no actual solution. I am trying to use a Cisco AS5300 as a > gateway for PSTNr. I have been able to configure it to take outbound > calls and send them to the PSTN just fine. Inbound calls however are > rejected by asterisk with "488 Not acceptable here" code. > > here are the details: > > AS5300: > IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE > SOFTWARE (fc5) > > Current configuration : 3939 bytes > > version 12.3 > service timestamps debug datetime msec > service timestamps log datetime msec > no service password-encryption > ! > hostname K_AS5300_3 > ! > boot-start-marker > boot-end-marker > ! > enable password ** > ! > resource-pool disable > clock timezone EST -5 > clock summer-time EDT recurring > ! > no aaa new-model > ip subnet-zero > ! > ! > isdn switch-type primary-dms100 > ! > ! > voice service voip > sip >bind all source-interface FastEthernet0 > > controller T1 0 > framing esf > clock source line primary > linecode b8zs > pri-group timeslots 1-24 > ! > controller T1 1 > framing esf > clock source line secondary 1 > linecode b8zs > pri-group timeslots 1-24 > ! > controller T1 2 > framing esf > linecode b8zs > pri-group timeslots 1-24 > ! > controller T1 3 > framing esf > linecode b8zs > pri-group timeslots 1-24 > ! > ! > ! > interface Ethernet0 > no ip address > shutdown > ! > interface Serial0:23 > no ip address > encapsulation hdlc > isdn switch-type primary-dms100 > isdn incoming-voice modem 64 > no cdp enable > ! > interface Serial1:23 > no ip address > encapsulation hdlc > isdn switch-type primary-dms100 > isdn incoming-voice modem 64 > no cdp enable > ! > interface Serial2:23 > no ip address > encapsulation hdlc > isdn switch-type primary-dms100 > isdn incoming-voice modem 64 > no cdp enable > ! > interface Serial3:23 > no ip address > encapsulation hdlc > isdn switch-type primary-dms100 > isdn incoming-voice modem 64 > no cdp enable > ! > interface FastEthernet0 > ip address 172.31.2.7 255.255.255.0 > duplex auto > speed auto > ! > ip classless > ip route 0.0.0.0 0.0.0.0 172.31.2.1 > no ip http server > ! > ! > ! > ! > ! > ! > voice-port 0:D > ! > voice-port 1:D > ! > voice-port 2:D > ! > voice-port 3:D > ! > ! > ! > dial-peer voice 100 voip > application session > destination-pattern 678... > signaling forward unconditional > session protocol sipv2 > session target sip-server > session transport udp > dtmf-relay rtp-nte > codec g711ulaw > no vad > ! > dial-peer voice 101 voip > destination-pattern 770... > progress_ind setup enable 3 > session protocol sipv2 > session target sip-server > session transport udp > dtmf-relay rtp-nte > codec g711ulaw > no vad > ! > dial-peer voice 102 voip > destination-pattern 404... > progress_ind setup enable 3 > session protocol sipv2 > session target sip-server > session transport udp > dtmf-relay rtp-nte > codec g711ulaw > no vad > ! > dial-peer voice 103 voip > destination-pattern 470... > progress_ind setup enable 3 > session protocol sipv2 > session target sip-server > session transport udp > dtmf-relay rtp-nte > codec g711ulaw > no vad > ! > dial-peer voice 200 pots > application session > incoming called-number . > destination-pattern 91.. > direct-inward-dial > port 0:D > prefix 1 > ! > dial-peer voice 201 pots > application session > incoming called-number . > destination-pattern 9.. > direct-inward-dial > port 0:D > ! > dial-peer voice 202 pots > application session > incoming called-number . > destination-pattern 91.. > direct-inward-dial > port 1:D > prefix 1 > ! > dial-peer voice 203 pots > application session > incoming called-number . > destination-pattern 9.. > direct-inward-dial > port 1:D > ! > dial-peer voice 204 pots > application session > incoming called-number . > destination-pattern 91.. > direct-inward-dial > dial-peer voice 204 pots > application session > incoming called-number . > destination-pattern 91.. > direct-inward-dial > port 2:D > prefix 1 > ! > dial-peer
[asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code
Hi I have searched the mailing lists and come across similar threads, but no actual solution. I am trying to use a Cisco AS5300 as a gateway for PSTNr. I have been able to configure it to take outbound calls and send them to the PSTN just fine. Inbound calls however are rejected by asterisk with "488 Not acceptable here" code. here are the details: AS5300: IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE SOFTWARE (fc5) Current configuration : 3939 bytes version 12.3 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname K_AS5300_3 ! boot-start-marker boot-end-marker ! enable password ** ! resource-pool disable clock timezone EST -5 clock summer-time EDT recurring ! no aaa new-model ip subnet-zero ! ! isdn switch-type primary-dms100 ! ! voice service voip sip bind all source-interface FastEthernet0 controller T1 0 framing esf clock source line primary linecode b8zs pri-group timeslots 1-24 ! controller T1 1 framing esf clock source line secondary 1 linecode b8zs pri-group timeslots 1-24 ! controller T1 2 framing esf linecode b8zs pri-group timeslots 1-24 ! controller T1 3 framing esf linecode b8zs pri-group timeslots 1-24 ! ! ! interface Ethernet0 no ip address shutdown ! interface Serial0:23 no ip address encapsulation hdlc isdn switch-type primary-dms100 isdn incoming-voice modem 64 no cdp enable ! interface Serial1:23 no ip address encapsulation hdlc isdn switch-type primary-dms100 isdn incoming-voice modem 64 no cdp enable ! interface Serial2:23 no ip address encapsulation hdlc isdn switch-type primary-dms100 isdn incoming-voice modem 64 no cdp enable ! interface Serial3:23 no ip address encapsulation hdlc isdn switch-type primary-dms100 isdn incoming-voice modem 64 no cdp enable ! interface FastEthernet0 ip address 172.31.2.7 255.255.255.0 duplex auto speed auto ! ip classless ip route 0.0.0.0 0.0.0.0 172.31.2.1 no ip http server ! ! ! ! ! ! voice-port 0:D ! voice-port 1:D ! voice-port 2:D ! voice-port 3:D ! ! ! dial-peer voice 100 voip application session destination-pattern 678... signaling forward unconditional session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 101 voip destination-pattern 770... progress_ind setup enable 3 session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 102 voip destination-pattern 404... progress_ind setup enable 3 session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 103 voip destination-pattern 470... progress_ind setup enable 3 session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 200 pots application session incoming called-number . destination-pattern 91.. direct-inward-dial port 0:D prefix 1 ! dial-peer voice 201 pots application session incoming called-number . destination-pattern 9.. direct-inward-dial port 0:D ! dial-peer voice 202 pots application session incoming called-number . destination-pattern 91.. direct-inward-dial port 1:D prefix 1 ! dial-peer voice 203 pots application session incoming called-number . destination-pattern 9.. direct-inward-dial port 1:D ! dial-peer voice 204 pots application session incoming called-number . destination-pattern 91.. direct-inward-dial dial-peer voice 204 pots application session incoming called-number . destination-pattern 91.. direct-inward-dial port 2:D prefix 1 ! dial-peer voice 205 pots application session incoming called-number . destination-pattern 9.. direct-inward-dial port 2:D ! dial-peer voice 206 pots application session incoming called-number . destination-pattern 91.. direct-inward-dial port 3:D prefix 1 ! dial-peer voice 207 pots application session incoming called-number . destination-pattern 9.. direct-inward-dial port 3:D ! sip-ua retry invite 4 retry response 3 retry bye 2 retry cancel 2 sip-server ipv4:172.31.2.29 ! ! line con 0 line aux 0 line vty 0 4 password login ! ntp clock-period 17179848 ntp peer 192.43.244.18 end Asterisk: Asterisk 1.2.12.1 on a x86_64 running Linux sip.conf: [general] context=default ; Default context for incoming calls bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls [as5300_1] type=peer host=172.31.2.7 permit=172.31.2.7/255.255.255.255 defaultip=172.31.2.7 disallow=all allow=ulaw allow=gsm allow=alaw nat=no canreinvite=yes dtmfmode=rfc2833 I have al
