I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway out there somewhere, I wouldn't be too proud to take a look at that either! Asterisk configs would be great too! Thanks, Wes
Yep. It will work just fine. Schochet, Wes wrote:>I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make >this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like >it should bee useful for something! > >I'm perfectly happy to do my homework, but also don't feel thee need to >reinvent the wheel! So, links with relevant info would be appreciated. If >there is a config for a 2621 being used as a gateway out there somewhere, I >wouldn't be too proud to take a look at that either! Asterisk configs would >be great too! > >Thanks, > >Wes >_______________________________________________ >--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 > >
I have a 2811 working as a SIP gateway. My IOS version is 12.3(11)T5. Looking through my config I notice: sip-ua sip-server ipv4:<asterisk server ip address> Everything else in the config file is for our h323 call manager gear. I can't remember if I needed to add the above line to make a sip server run on the router. In order to place a call to the PSTN, I Dial(SIP/9XXXXXXXXXX@<ip address of my 2811>) and everything works. As for how much of this applies to a 2600.. you'll have to see. On 2/6/06, Schochet, Wes <wes.schochet@selectcomfort.com> wrote:> > I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make > this thing into MGCP gateway or even a SIP gateway for asterisk? Seems > like > it should bee useful for something! > > I'm perfectly happy to do my homework, but also don't feel thee need to > reinvent the wheel! So, links with relevant info would be > appreciated. If > there is a config for a 2621 being used as a gateway out there somewhere, > I > wouldn't be too proud to take a look at that either! Asterisk configs > would > be great too! > > Thanks, > > Wes > _______________________________________________ > --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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060207/4afec5c0/attachment.htm
Gary- You have a Cisco 2600 acting as both a SIP gateway to the Asterisk box -and- as an H.323 gateway for your CCM? Can you provide me with config details? 12.3(8)T3,c2600-ipvoice-mz.123-8.T3 with T1 (2 Port) Multi-Flex Trunk I tried to set up h.323 to the Asterisk box-- didn't know there was a possibility of running SIP and H.323 at the same time... thanks, Tim ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gary Richardson Sent: Tuesday, February 07, 2006 9:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 2620 as PRI gateway I have a 2811 working as a SIP gateway. My IOS version is 12.3(11)T5. Looking through my config I notice: sip-ua sip-server ipv4:<asterisk server ip address> Everything else in the config file is for our h323 call manager gear. I can't remember if I needed to add the above line to make a sip server run on the router. In order to place a call to the PSTN, I Dial(SIP/9XXXXXXXXXX@<ip address of my 2811>) and everything works. As for how much of this applies to a 2600.. you'll have to see. On 2/6/06, Schochet, Wes <wes.schochet@selectcomfort.com > wrote: I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway out there somewhere, I wouldn't be too proud to take a look at that either! Asterisk configs would be great too! Thanks, Wes _______________________________________________ --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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060208/3f7ce7db/attachment.htm
sip-ua sip-server ipv4:<asterisk server ip address> OK - So I added those lines to my 2651 with the IP of my asterisk box... How would I set this up as a SIP trunk in Asterisk? I have done this, in building a SIP trunk in AMP. host=10.12.1.252 type=friend I don't know if/how to specify a username/password (as was the defaults in there- the router didn't support having that configured..) So I picked friend.. Then, in call routing, I picked my "Outbound Routing" the "9_outside" route of "9|." Set that to use the new 'gw-rtr' I'd created... no go... Debug ISDN q931 doesn't show anything going to the router... In Asterisk- " -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 10.12.1.252" <snipped from below> The router doesn't show anything... the below shows up in Asterisk -vvvv mode -- Executing Macro("SIP/6351-cc18", "dialout-trunk|3|2439499|") in new stack -- Executing GotoIf("SIP/6351-cc18", "1?3:2)") in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro("SIP/6351-cc18", "user-callerid") in new stack -- Executing DBget("SIP/6351-cc18", "AMPUSER=DEVICE/6351/user") in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=6351/user -- DBget: set variable AMPUSER to 6351 -- Executing DBget("SIP/6351-cc18", "AMPUSERCIDNAME=AMPUSER/6351/cidname") in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=6351/cidname -- DBget: set variable AMPUSERCIDNAME to Tim-Zyxel -- Executing GotoIf("SIP/6351-cc18", "0?5") in new stack -- Executing SetCallerID("SIP/6351-cc18", "Tim-Zyxel <6351>") in new stack -- Executing NoOp("SIP/6351-cc18", "Using CallerID "Tim-Zyxel" <6351>") in new stack -- Executing Macro("SIP/6351-cc18", "record-enable|6351|OUT") in new stack -- Executing GotoIf("SIP/6351-cc18", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/6351-cc18", "recordingcheck|20060208-115748|1139417868.14") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060208-115748|1139417868.14: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/6351-cc18", "No recording needed") in new stack -- Executing Macro("SIP/6351-cc18", "outbound-callerid|3") in new stack -- Executing GotoIf("SIP/6351-cc18", "1?3") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing DBget("SIP/6351-cc18", "USEROUTCID=AMPUSER/6351/outboundcid") in new stack -- DBget: varname=USEROUTCID, family=AMPUSER, key=6351/outboundcid -- DBget: set variable USEROUTCID to 6351 -- Executing GotoIf("SIP/6351-cc18", "0?6") in new stack -- Executing SetCallerID("SIP/6351-cc18", "6351") in new stack -- Executing NoOp("SIP/6351-cc18", "CallerID set to 6351") in new stack -- Executing SetGroup("SIP/6351-cc18", "OUT_3") in new stack -- Executing CheckGroup("SIP/6351-cc18", "") in new stack -- Executing SetVar("SIP/6351-cc18", "DIAL_NUMBER=2439499") in new stack -- Executing SetVar("SIP/6351-cc18", "DIAL_TRUNK=3") in new stack -- Executing AGI("SIP/6351-cc18", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar("SIP/6351-cc18", "OUTNUM=2439499") in new stack -- Executing Cut("SIP/6351-cc18", "custom=OUT_3|:|1") in new stack -- Executing GotoIf("SIP/6351-cc18", "0?16") in new stack -- Executing Dial("SIP/6351-cc18", "SIP/acs-gw-rtr/2439499") in new stack -- Called acs-gw-rtr/2439499 -- SIP/acs-gw-rtr-b33f is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto("SIP/6351-cc18", "s-CONGESTION|1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing NoOp("SIP/6351-cc18", "Dial failed due to CONGESTION") in new stack -- Executing Macro("SIP/6351-cc18", "outisbusy") in new stack -- Executing Playback("SIP/6351-cc18", "allison7/all-circuits-busy-now") in new stack -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 10.12.1.252 -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback("SIP/6351-cc18", "allison7/pls-try-call-later") in new stack -- Playing 'allison7/pls-try-call-later' (language 'en') -- Executing Macro("SIP/6351-cc18", "hangupcall") in new stack -- Executing ResetCDR("SIP/6351-cc18", "w") in new stack -- Executing NoCDR("SIP/6351-cc18", "") in new stack -- Executing Wait("SIP/6351-cc18", "5") in new stack -- Executing Hangup("SIP/6351-cc18", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/6351-cc18' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/6351-cc18' in macro 'outisbusy' == Spawn extension (from-internal, 92439499, 2) exited non-zero on 'SIP/6351-cc18' -- Executing Macro("SIP/6351-cc18", "hangupcall") in new stack -- Executing ResetCDR("SIP/6351-cc18", "w") in new stack -- Executing NoCDR("SIP/6351-cc18", "") in new stack -- Executing Wait("SIP/6351-cc18", "5") in new stack -- Executing Hangup("SIP/6351-cc18", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/6351-cc18' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/6351-cc18' acsasterisk*CLI> ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gary Richardson Sent: Tuesday, February 07, 2006 9:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 2620 as PRI gateway I have a 2811 working as a SIP gateway. My IOS version is 12.3(11)T5. Looking through my config I notice: sip-ua sip-server ipv4:<asterisk server ip address> Everything else in the config file is for our h323 call manager gear. I can't remember if I needed to add the above line to make a sip server run on the router. In order to place a call to the PSTN, I Dial(SIP/9XXXXXXXXXX@<ip address of my 2811>) and everything works. As for how much of this applies to a 2600.. you'll have to see. On 2/6/06, Schochet, Wes <wes.schochet@selectcomfort.com > wrote: I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway out there somewhere, I wouldn't be too proud to take a look at that either! Asterisk configs would be great too! Thanks, Wes _______________________________________________ --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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060208/2d99acb8/attachment.htm
Did you create the dial-peers in the2651? -----Mensaje original----- De: Tim Reimers [mailto:tim.reimers@asheville.k12.nc.us] Enviado el: Wednesday, February 08, 2006 1:41 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] Cisco 2620 as PRI gateway sip-ua sip-server ipv4:<asterisk server ip address> OK - So I added those lines to my 2651 with the IP of my asterisk box... How would I set this up as a SIP trunk in Asterisk? I have done this, in building a SIP trunk in AMP. host=10.12.1.252 type=friend I don't know if/how to specify a username/password (as was the defaults in there- the router didn't support having that configured..) So I picked friend.. Then, in call routing, I picked my "Outbound Routing" the "9_outside" route of "9|." Set that to use the new 'gw-rtr' I'd created... no go... Debug ISDN q931 doesn't show anything going to the router... In Asterisk- " -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 10.12.1.252" <snipped from below> The router doesn't show anything... the below shows up in Asterisk -vvvv mode -- Executing Macro("SIP/6351-cc18", "dialout-trunk|3|2439499|") in new stack -- Executing GotoIf("SIP/6351-cc18", "1?3:2)") in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro("SIP/6351-cc18", "user-callerid") in new stack -- Executing DBget("SIP/6351-cc18", "AMPUSER=DEVICE/6351/user") in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=6351/user -- DBget: set variable AMPUSER to 6351 -- Executing DBget("SIP/6351-cc18", "AMPUSERCIDNAME=AMPUSER/6351/cidname") in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=6351/cidname -- DBget: set variable AMPUSERCIDNAME to Tim-Zyxel -- Executing GotoIf("SIP/6351-cc18", "0?5") in new stack -- Executing SetCallerID("SIP/6351-cc18", "Tim-Zyxel <6351>") in new stack -- Executing NoOp("SIP/6351-cc18", "Using CallerID "Tim-Zyxel" <6351>") in new stack -- Executing Macro("SIP/6351-cc18", "record-enable|6351|OUT") in new stack -- Executing GotoIf("SIP/6351-cc18", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/6351-cc18", "recordingcheck|20060208-115748|1139417868.14") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060208-115748|1139417868.14: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/6351-cc18", "No recording needed") in new stack -- Executing Macro("SIP/6351-cc18", "outbound-callerid|3") in new stack -- Executing GotoIf("SIP/6351-cc18", "1?3") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing DBget("SIP/6351-cc18", "USEROUTCID=AMPUSER/6351/outboundcid") in new stack -- DBget: varname=USEROUTCID, family=AMPUSER, key=6351/outboundcid -- DBget: set variable USEROUTCID to 6351 -- Executing GotoIf("SIP/6351-cc18", "0?6") in new stack -- Executing SetCallerID("SIP/6351-cc18", "6351") in new stack -- Executing NoOp("SIP/6351-cc18", "CallerID set to 6351") in new stack -- Executing SetGroup("SIP/6351-cc18", "OUT_3") in new stack -- Executing CheckGroup("SIP/6351-cc18", "") in new stack -- Executing SetVar("SIP/6351-cc18", "DIAL_NUMBER=2439499") in new stack -- Executing SetVar("SIP/6351-cc18", "DIAL_TRUNK=3") in new stack -- Executing AGI("SIP/6351-cc18", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar("SIP/6351-cc18", "OUTNUM=2439499") in new stack -- Executing Cut("SIP/6351-cc18", "custom=OUT_3|:|1") in new stack -- Executing GotoIf("SIP/6351-cc18", "0?16") in new stack -- Executing Dial("SIP/6351-cc18", "SIP/acs-gw-rtr/2439499") in new stack -- Called acs-gw-rtr/2439499 -- SIP/acs-gw-rtr-b33f is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto("SIP/6351-cc18", "s-CONGESTION|1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing NoOp("SIP/6351-cc18", "Dial failed due to CONGESTION") in new stack -- Executing Macro("SIP/6351-cc18", "outisbusy") in new stack -- Executing Playback("SIP/6351-cc18", "allison7/all-circuits-busy-now") in new stack -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 10.12.1.252 -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback("SIP/6351-cc18", "allison7/pls-try-call-later") in new stack -- Playing 'allison7/pls-try-call-later' (language 'en') -- Executing Macro("SIP/6351-cc18", "hangupcall") in new stack -- Executing ResetCDR("SIP/6351-cc18", "w") in new stack -- Executing NoCDR("SIP/6351-cc18", "") in new stack -- Executing Wait("SIP/6351-cc18", "5") in new stack -- Executing Hangup("SIP/6351-cc18", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/6351-cc18' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/6351-cc18' in macro 'outisbusy' == Spawn extension (from-internal, 92439499, 2) exited non-zero on 'SIP/6351-cc18' -- Executing Macro("SIP/6351-cc18", "hangupcall") in new stack -- Executing ResetCDR("SIP/6351-cc18", "w") in new stack -- Executing NoCDR("SIP/6351-cc18", "") in new stack -- Executing Wait("SIP/6351-cc18", "5") in new stack -- Executing Hangup("SIP/6351-cc18", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/6351-cc18' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/6351-cc18' acsasterisk*CLI> _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gary Richardson Sent: Tuesday, February 07, 2006 9:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 2620 as PRI gateway I have a 2811 working as a SIP gateway. My IOS version is 12.3(11)T5. Looking through my config I notice: sip-ua sip-server ipv4:<asterisk server ip address> Everything else in the config file is for our h323 call manager gear. I can't remember if I needed to add the above line to make a sip server run on the router. In order to place a call to the PSTN, I Dial(SIP/9XXXXXXXXXX@<ip address of my 2811>) and everything works. As for how much of this applies to a 2600.. you'll have to see. On 2/6/06, Schochet, Wes < wes.schochet@selectcomfort.com <mailto:wes.schochet@selectcomfort.com> > wrote: I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway out there somewhere, I wouldn't be too proud to take a look at that either! Asterisk configs would be great too! Thanks, Wes _______________________________________________ --Bandwidth and Colocation provided by Easynews.com <http://Easynews.com> -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060208/8a6c3df0/attachment.htm
Yeah-- sorry... " dial-peer voice 635099 voip description calls sent to Asterisk preference 1 destination-pattern [635-9].. progress_ind setup enable 3 session target ipv4:10.10.1.28 dtmf-relay h245-alphanumeric " I had been trying to do this with H.323 -- the Call Manager uses H.323 There are some sip commands available in that dial-peer ACS-GW(config-dial-peer)#voice-class sip ? rel1xx Type of reliable provisional response support transport Configure transport related parameters url url type in request line of outgoing INVITE Not sure how I set those--- This: voice-class codec 1 voice-class h323 1 is what is in there for the Call Manager h.323 dial-peer That's obviously NOT what I want for the Asterisk-SIP connection... but I don't know what I need to do regarding the 'sip url' or 'sip transport' or 'sip rel1xx' commands, if anything... How does one debug SIP activity? I see the debugs for calls--- but I don't know the related debugs for actively watching-- like you would 'debug isdn q931' -- that's the outgoing side of the router-- what would be the debug for a SIP call 'arriving' at the router?? ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Salas Sent: Wednesday, February 08, 2006 2:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 2620 as PRI gateway Did you create the dial-peers in the2651? -----Mensaje original----- De: Tim Reimers [mailto:tim.reimers@asheville.k12.nc.us] Enviado el: Wednesday, February 08, 2006 1:41 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] Cisco 2620 as PRI gateway sip-ua sip-server ipv4:<asterisk server ip address> OK - So I added those lines to my 2651 with the IP of my asterisk box... How would I set this up as a SIP trunk in Asterisk? I have done this, in building a SIP trunk in AMP. host=10.12.1.252 type=friend I don't know if/how to specify a username/password (as was the defaults in there- the router didn't support having that configured..) So I picked friend.. Then, in call routing, I picked my "Outbound Routing" the "9_outside" route of "9|." Set that to use the new 'gw-rtr' I'd created... no go... Debug ISDN q931 doesn't show anything going to the router... In Asterisk- " -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 10.12.1.252" <snipped from below> The router doesn't show anything... the below shows up in Asterisk -vvvv mode -- Executing Macro("SIP/6351-cc18", "dialout-trunk|3|2439499|") in new stack -- Executing GotoIf("SIP/6351-cc18", "1?3:2)") in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro("SIP/6351-cc18", "user-callerid") in new stack -- Executing DBget("SIP/6351-cc18", "AMPUSER=DEVICE/6351/user") in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=6351/user -- DBget: set variable AMPUSER to 6351 -- Executing DBget("SIP/6351-cc18", "AMPUSERCIDNAME=AMPUSER/6351/cidname") in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=6351/cidname -- DBget: set variable AMPUSERCIDNAME to Tim-Zyxel -- Executing GotoIf("SIP/6351-cc18", "0?5") in new stack -- Executing SetCallerID("SIP/6351-cc18", "Tim-Zyxel <6351>") in new stack -- Executing NoOp("SIP/6351-cc18", "Using CallerID "Tim-Zyxel" <6351>") in new stack -- Executing Macro("SIP/6351-cc18", "record-enable|6351|OUT") in new stack -- Executing GotoIf("SIP/6351-cc18", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/6351-cc18", "recordingcheck|20060208-115748|1139417868.14") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060208-115748|1139417868.14: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/6351-cc18", "No recording needed") in new stack -- Executing Macro("SIP/6351-cc18", "outbound-callerid|3") in new stack -- Executing GotoIf("SIP/6351-cc18", "1?3") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing DBget("SIP/6351-cc18", "USEROUTCID=AMPUSER/6351/outboundcid") in new stack -- DBget: varname=USEROUTCID, family=AMPUSER, key=6351/outboundcid -- DBget: set variable USEROUTCID to 6351 -- Executing GotoIf("SIP/6351-cc18", "0?6") in new stack -- Executing SetCallerID("SIP/6351-cc18", "6351") in new stack -- Executing NoOp("SIP/6351-cc18", "CallerID set to 6351") in new stack -- Executing SetGroup("SIP/6351-cc18", "OUT_3") in new stack -- Executing CheckGroup("SIP/6351-cc18", "") in new stack -- Executing SetVar("SIP/6351-cc18", "DIAL_NUMBER=2439499") in new stack -- Executing SetVar("SIP/6351-cc18", "DIAL_TRUNK=3") in new stack -- Executing AGI("SIP/6351-cc18", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar("SIP/6351-cc18", "OUTNUM=2439499") in new stack -- Executing Cut("SIP/6351-cc18", "custom=OUT_3|:|1") in new stack -- Executing GotoIf("SIP/6351-cc18", "0?16") in new stack -- Executing Dial("SIP/6351-cc18", "SIP/acs-gw-rtr/2439499") in new stack -- Called acs-gw-rtr/2439499 -- SIP/acs-gw-rtr-b33f is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto("SIP/6351-cc18", "s-CONGESTION|1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing NoOp("SIP/6351-cc18", "Dial failed due to CONGESTION") in new stack -- Executing Macro("SIP/6351-cc18", "outisbusy") in new stack -- Executing Playback("SIP/6351-cc18", "allison7/all-circuits-busy-now") in new stack -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 10.12.1.252 -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback("SIP/6351-cc18", "allison7/pls-try-call-later") in new stack -- Playing 'allison7/pls-try-call-later' (language 'en') -- Executing Macro("SIP/6351-cc18", "hangupcall") in new stack -- Executing ResetCDR("SIP/6351-cc18", "w") in new stack -- Executing NoCDR("SIP/6351-cc18", "") in new stack -- Executing Wait("SIP/6351-cc18", "5") in new stack -- Executing Hangup("SIP/6351-cc18", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/6351-cc18' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/6351-cc18' in macro 'outisbusy' == Spawn extension (from-internal, 92439499, 2) exited non-zero on 'SIP/6351-cc18' -- Executing Macro("SIP/6351-cc18", "hangupcall") in new stack -- Executing ResetCDR("SIP/6351-cc18", "w") in new stack -- Executing NoCDR("SIP/6351-cc18", "") in new stack -- Executing Wait("SIP/6351-cc18", "5") in new stack -- Executing Hangup("SIP/6351-cc18", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/6351-cc18' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/6351-cc18' acsasterisk*CLI> ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gary Richardson Sent: Tuesday, February 07, 2006 9:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 2620 as PRI gateway I have a 2811 working as a SIP gateway. My IOS version is 12.3(11)T5. Looking through my config I notice: sip-ua sip-server ipv4:<asterisk server ip address> Everything else in the config file is for our h323 call manager gear. I can't remember if I needed to add the above line to make a sip server run on the router. In order to place a call to the PSTN, I Dial(SIP/9XXXXXXXXXX@<ip address of my 2811>) and everything works. As for how much of this applies to a 2600.. you'll have to see. On 2/6/06, Schochet, Wes <wes.schochet@selectcomfort.com > wrote: I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway out there somewhere, I wouldn't be too proud to take a look at that either! Asterisk configs would be great too! Thanks, Wes _______________________________________________ --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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060209/c94bd489/attachment.htm
If you are using the 2620 like a SIP IP-PSTN gateway your voip dial-peer would be like this: dial-peer voice 635099 voip description calls sent to Asterisk preference 1 destination-pattern T (or whatever you need to match) session target sip-server dtmf-relay h245-alphanumeric (or whatever you need) session-protocol sip no vad And you need a pots dial-peer, something like this dial-peer voice 0 pots destination-pattern T (or whatever you need) port 0/0 0 And in sip-ua: sip-ua sip-server <asterisk server ip address> This is the basic Regards Jsalas -----Mensaje original----- De: Tim Reimers [mailto:tim.reimers@asheville.k12.nc.us] Enviado el: Thursday, February 09, 2006 10:04 AM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] Cisco 2620 as PRI gateway Yeah-- sorry... " dial-peer voice 635099 voip description calls sent to Asterisk preference 1 destination-pattern [635-9].. progress_ind setup enable 3 session target ipv4:10.10.1.28 dtmf-relay h245-alphanumeric " I had been trying to do this with H.323 -- the Call Manager uses H.323 There are some sip commands available in that dial-peer ACS-GW(config-dial-peer)#voice-class sip ? rel1xx Type of reliable provisional response support transport Configure transport related parameters url url type in request line of outgoing INVITE Not sure how I set those--- This: voice-class codec 1 voice-class h323 1 is what is in there for the Call Manager h.323 dial-peer That's obviously NOT what I want for the Asterisk-SIP connection... but I don't know what I need to do regarding the 'sip url' or 'sip transport' or 'sip rel1xx' commands, if anything... How does one debug SIP activity? I see the debugs for calls--- but I don't know the related debugs for actively watching-- like you would 'debug isdn q931' -- that's the outgoing side of the router-- what would be the debug for a SIP call 'arriving' at the router?? _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Salas Sent: Wednesday, February 08, 2006 2:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 2620 as PRI gateway Did you create the dial-peers in the2651? -----Mensaje original----- De: Tim Reimers [mailto:tim.reimers@asheville.k12.nc.us] Enviado el: Wednesday, February 08, 2006 1:41 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] Cisco 2620 as PRI gateway sip-ua sip-server ipv4:<asterisk server ip address> OK - So I added those lines to my 2651 with the IP of my asterisk box... How would I set this up as a SIP trunk in Asterisk? I have done this, in building a SIP trunk in AMP. host=10.12.1.252 type=friend I don't know if/how to specify a username/password (as was the defaults in there- the router didn't support having that configured..) So I picked friend.. Then, in call routing, I picked my "Outbound Routing" the "9_outside" route of "9|." Set that to use the new 'gw-rtr' I'd created... no go... Debug ISDN q931 doesn't show anything going to the router... In Asterisk- " -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 10.12.1.252" <snipped from below> The router doesn't show anything... the below shows up in Asterisk -vvvv mode -- Executing Macro("SIP/6351-cc18", "dialout-trunk|3|2439499|") in new stack -- Executing GotoIf("SIP/6351-cc18", "1?3:2)") in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro("SIP/6351-cc18", "user-callerid") in new stack -- Executing DBget("SIP/6351-cc18", "AMPUSER=DEVICE/6351/user") in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=6351/user -- DBget: set variable AMPUSER to 6351 -- Executing DBget("SIP/6351-cc18", "AMPUSERCIDNAME=AMPUSER/6351/cidname") in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=6351/cidname -- DBget: set variable AMPUSERCIDNAME to Tim-Zyxel -- Executing GotoIf("SIP/6351-cc18", "0?5") in new stack -- Executing SetCallerID("SIP/6351-cc18", "Tim-Zyxel <6351>") in new stack -- Executing NoOp("SIP/6351-cc18", "Using CallerID "Tim-Zyxel" <6351>") in new stack -- Executing Macro("SIP/6351-cc18", "record-enable|6351|OUT") in new stack -- Executing GotoIf("SIP/6351-cc18", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/6351-cc18", "recordingcheck|20060208-115748|1139417868.14") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060208-115748|1139417868.14: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/6351-cc18", "No recording needed") in new stack -- Executing Macro("SIP/6351-cc18", "outbound-callerid|3") in new stack -- Executing GotoIf("SIP/6351-cc18", "1?3") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing DBget("SIP/6351-cc18", "USEROUTCID=AMPUSER/6351/outboundcid") in new stack -- DBget: varname=USEROUTCID, family=AMPUSER, key=6351/outboundcid -- DBget: set variable USEROUTCID to 6351 -- Executing GotoIf("SIP/6351-cc18", "0?6") in new stack -- Executing SetCallerID("SIP/6351-cc18", "6351") in new stack -- Executing NoOp("SIP/6351-cc18", "CallerID set to 6351") in new stack -- Executing SetGroup("SIP/6351-cc18", "OUT_3") in new stack -- Executing CheckGroup("SIP/6351-cc18", "") in new stack -- Executing SetVar("SIP/6351-cc18", "DIAL_NUMBER=2439499") in new stack -- Executing SetVar("SIP/6351-cc18", "DIAL_TRUNK=3") in new stack -- Executing AGI("SIP/6351-cc18", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar("SIP/6351-cc18", "OUTNUM=2439499") in new stack -- Executing Cut("SIP/6351-cc18", "custom=OUT_3|:|1") in new stack -- Executing GotoIf("SIP/6351-cc18", "0?16") in new stack -- Executing Dial("SIP/6351-cc18", "SIP/acs-gw-rtr/2439499") in new stack -- Called acs-gw-rtr/2439499 -- SIP/acs-gw-rtr-b33f is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto("SIP/6351-cc18", "s-CONGESTION|1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing NoOp("SIP/6351-cc18", "Dial failed due to CONGESTION") in new stack -- Executing Macro("SIP/6351-cc18", "outisbusy") in new stack -- Executing Playback("SIP/6351-cc18", "allison7/all-circuits-busy-now") in new stack -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 10.12.1.252 -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback("SIP/6351-cc18", "allison7/pls-try-call-later") in new stack -- Playing 'allison7/pls-try-call-later' (language 'en') -- Executing Macro("SIP/6351-cc18", "hangupcall") in new stack -- Executing ResetCDR("SIP/6351-cc18", "w") in new stack -- Executing NoCDR("SIP/6351-cc18", "") in new stack -- Executing Wait("SIP/6351-cc18", "5") in new stack -- Executing Hangup("SIP/6351-cc18", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/6351-cc18' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/6351-cc18' in macro 'outisbusy' == Spawn extension (from-internal, 92439499, 2) exited non-zero on 'SIP/6351-cc18' -- Executing Macro("SIP/6351-cc18", "hangupcall") in new stack -- Executing ResetCDR("SIP/6351-cc18", "w") in new stack -- Executing NoCDR("SIP/6351-cc18", "") in new stack -- Executing Wait("SIP/6351-cc18", "5") in new stack -- Executing Hangup("SIP/6351-cc18", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/6351-cc18' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/6351-cc18' acsasterisk*CLI> _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gary Richardson Sent: Tuesday, February 07, 2006 9:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 2620 as PRI gateway I have a 2811 working as a SIP gateway. My IOS version is 12.3(11)T5. Looking through my config I notice: sip-ua sip-server ipv4:<asterisk server ip address> Everything else in the config file is for our h323 call manager gear. I can't remember if I needed to add the above line to make a sip server run on the router. In order to place a call to the PSTN, I Dial(SIP/9XXXXXXXXXX@<ip address of my 2811>) and everything works. As for how much of this applies to a 2600.. you'll have to see. On 2/6/06, Schochet, Wes < wes.schochet@selectcomfort.com <mailto:wes.schochet@selectcomfort.com> > wrote: I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway out there somewhere, I wouldn't be too proud to take a look at that either! Asterisk configs would be great too! Thanks, Wes _______________________________________________ --Bandwidth and Colocation provided by Easynews.com <http://Easynews.com> -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060209/d4a0fd69/attachment.htm
debug ccsip message Kurt
Thanks! -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of kurt x Sent: Friday, February 10, 2006 10:51 AM To: Asterisk Subject: RE: [Asterisk-Users] Cisco 2620 as PRI gateway debug ccsip message Kurt _______________________________________________ --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
Here's a debug--- Longish, but I'm not sure what info in this might be useful to anyone-- I have a zyxel SIP phone configured as ext '6351' on Asterisk-- I can successfully call the SIP phone from an Sjphone client on my PC and talk between the two- so the SIP phone is in fact registered with * correctly... The Cisco router has matching for 63[5-9]x configured- Here's a debug from the router: ACS-GW# *May 26 13:02:42: ISDN Se1/1:23 Q931: pak_private_number: Invalid type/plan 0x0 0x1 may be overriden; sw-type 13 *May 26 13:02:42: ISDN Se1/1:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13 *May 26 13:02:42: ISDN Se1/1:23 Q931: Applying typeplan for sw-type 0xD is 0x4 0 x1, Called num 3506351 *May 26 13:02:42: ISDN Se1/1:23 Q931: TX -> SETUP pd = 8 callref 0x7A54 Bearer Capability i = 0x8090A2 Standard = CCITT Transer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA18387 Preferred, Channel 7 Calling Party Number i = 0x2181, '8283506180' Plan:ISDN, Type:National Called Party Number i = 0xC1, '3506351' Plan:ISDN, Type:Subscriber(local) *May 26 13:02:42: ISDN Se1/1:23 Q931: RX <- CALL_PROC pd = 8 callref 0xFA54 Channel ID i = 0xA98387 Exclusive, Channel 7 *May 26 13:02:43: ISDN Se1/0:23 Q931: RX <- SETUP pd = 8 callref 0x0014 Bearer Capability i = 0x8090A2 Standard = CCITT Transer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Calling Party Number i = 0x2181, '8283506180' Plan:ISDN, Type:National Called Party Number i = 0x80, '6351' Plan:Unknown, Type:Unknown *May 26 13:02:43: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:6351@10.10.1.28:5060 SIP/2.0 Via: SIP/2.0/UDP 10.12.1.252:5060;branch=z9hG4bK12A7 From: <sip:8283506180@10.12.1.252>;tag=3FB70415-7B4 To: <sip:6351@10.10.1.28> Date: Sun, 26 May 2002 18:02:43 gmt Call-ID: 9C227F6C-700911D6-803FEF66-8E03C25C@10.12.1.252 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 2619306202-1879642582-3189047311-607696096 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY , INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: <sip:8283506180@10.12.1.252>;party=calling;screen=yes;privacy=o ff Timestamp: 1022436163 Contact: <sip:8283506180@10.12.1.252:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 211 v=0 o=CiscoSystemsSIP-GW-UserAgent 7318 2409 IN IP4 10.12.1.252 s=SIP Call c=IN IP4 10.12.1.252 t=0 0 m=audio 16396 RTP/AVP 18 c=IN IP4 10.12.1.252 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 *May 26 13:02:43: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 10.12.1.252:5060;branch=z9hG4bK12A7 From: <sip:8283506180@10.12.1.252>;tag=3FB70415-7B4 To: <sip:6351@10.10.1.28>;tag=as21b1fb71 Call-ID: 9C227F6C-700911D6-803FEF66-8E03C25C@10.12.1.252 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:6351@10.10.1.28> Content-Length: 0 *May 26 13:02:43: ISDN Se1/0:23 Q931: TX -> CALL_PROC pd = 8 callref 0x8014 Channel ID i = 0xA18383 Preferred, Channel 3 ACS-GW# *May 26 13:02:43: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:6351@10.10.1.28:5060 SIP/2.0 Via: SIP/2.0/UDP 10.12.1.252:5060;branch=z9hG4bK12A7 From: <sip:8283506180@10.12.1.252>;tag=3FB70415-7B4 To: <sip:6351@10.10.1.28>;tag=as21b1fb71 Date: Sun, 26 May 2002 18:02:43 gmt Call-ID: 9C227F6C-700911D6-803FEF66-8E03C25C@10.12.1.252 Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 *May 26 13:02:43: ISDN Se1/0:23 Q931: TX -> DISCONNECT pd = 8 callref 0x8014 Cause i = 0x8081 - Unallocated/unassigned number ACS-GW# *May 26 13:02:43: ISDN Se1/1:23 Q931: RX <- PROGRESS pd = 8 callref 0xFA54 Cause i = 0x8281 - Unallocated/unassigned number Progress Ind i = 0x8288 - In-band info or appropriate now available *May 26 13:02:43: ISDN Se1/0:23 Q931: RX <- RELEASE pd = 8 callref 0x0014 *May 26 13:02:43: ISDN Se1/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x801 4 ACS-GW# *May 26 13:02:49: ISDN Se1/1:23 Q931: TX -> DISCONNECT pd = 8 callref 0x7A54 Cause i = 0x8090 - Normal call clearing *May 26 13:02:49: ISDN Se1/1:23 Q931: RX <- RELEASE pd = 8 callref 0xFA54 *May 26 13:02:49: ISDN Se1/1:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x7A5 4 ACS-GW# *May 26 13:03:07: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: OPTIONS sip:10.12.1.252 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.28:5060;branch=z9hG4bK3741dad8 From: "Unknown" <sip:Unknown@10.10.1.28>;tag=as6479f479 To: <sip:10.12.1.252> Contact: <sip:Unknown@10.10.1.28> Call-ID: 6668302302bd600a6cdd9b830991189e@10.10.1.28 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 13 Feb 2006 18:58:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 *May 26 13:03:07: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.28:5060;branch=z9hG4bK3741dad8 From: "Unknown" <sip:Unknown@10.10.1.28>;tag=as6479f479 To: <sip:10.12.1.252>;tag=3FB76385-1200 Date: Sun, 26 May 2002 18:03:07 gmt Call-ID: 6668302302bd600a6cdd9b830991189e@10.10.1.28 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 OPTIONS Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY , INFO, UPDATE, REGISTER Accept: application/sdp Allow-Events: telephone-event Content-Length: 163 Content-Type: ACS-GW# application/sdp v=0 o=CiscoSystemsSIP-GW-UserAgent 777 2006 IN IP4 10.12.1.252 s=SIP Call c=IN IP4 10.12.1.252 t=0 0 m=audio 0 RTP/AVP 18 0 8 4 2 15 3 c=IN IP4 10.12.1.252 Asterisk shows nothing in '-r -vvvv' mode.. Anywhere I can trace to see if * is actually taking the call at all? Here's this--- it LOOKS like the router is known to the * server.... acsasterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status pots 10.12.1.252 255.255.255.255 5060 OK (47 ms) callmgr 10.12.1.11 255.255.255.255 5060 Unmonitored acs-gw-rtr 10.12.1.252 255.255.255.255 5060 Unmonitored 6399/6399 (Unspecified) D 255.255.255.255 0 Unmonitored 6352/6352 10.10.1.183 D 255.255.255.255 5060 Unmonitored 6351/6351 10.10.1.101 D 255.255.255.255 5060 Unmonitored 6 sip peers [6 online , 0 offline] acsasterisk*CLI> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of kurt x Sent: Friday, February 10, 2006 10:51 AM To: Asterisk Subject: RE: [Asterisk-Users] Cisco 2620 as PRI gateway debug ccsip message Kurt _______________________________________________ --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
A 488 can mean a codec miss match. Check that your Asterisk box is configured for g729. Kurt
That worked! thanks Kurt... On a side note... Somehow, ALL of the phone calls ended up coming to my SIP phone--- I need to figure out the appropriate 'inbound routing' such that all calls coming from the PRI router (extensions 6350 through 6399) get sent directly to the right extension... right now, the incoming routing is set to 'use incoming calls' settings... I don't know what I need to configure such that calls for all 50 DIDs arriving 'from' the router are handled by (hopefully one route) that says "send the call to the extension that matches the digits" Thanks, Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of kurt x Sent: Tuesday, February 14, 2006 9:50 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Cisco 2620 as PRI gateway A 488 can mean a codec miss match. Check that your Asterisk box is configured for g729. Kurt _______________________________________________ --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