[Asterisk-Users] Cisco + FXO PORT
Has anyone successful setup up asterisk with cisco router with FXO ___ --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] Cisco FXO hangup detection
I am using a Cisco 1760V with FXO card in Australia to provide ports into Asterisk. I was wondering if anyone out there has a config for the cisco to detect the disconnect or hangup signal for Australian tones. If the calling party hangs up while leaving a voice mail for example, it takes around 15 seconds for the call to time out. I believe the Cisco can be configured to detect the hangup or disconnect tone, but l can't find any details in my searching. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco FXO Caller-ID
When caller-id enabled on voice port 2/3 system plays a congestion tone on answer (call never makes it to the Asterisk server), when caller-id is disabled, everything works fine. System is a Cisco 1760 running 12.3(10) on a VIC2-4FXO. dial-peer voice 2 voip description Route Calls to Asterisk destination-pattern 790 session protocol sipv2 session target sip-server dtmf-relay rtp-nte codec g711ulaw no vad sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:10.0.1.129 voice-port 2/3 no comfort-noise connection plar 790 caller-id enable Debugging shows (when caller ID is enabled): htsp_pre_connect_disconnect, cdb = 821DB748 cause = 11 Any help would be appreciated! ___ 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
RE: [Asterisk-Users] Cisco FXO
You need to upgrade to the latest IOS. IP voice or IP plus for full SIP features. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Espinoza Sent: Wednesday, October 13, 2004 8:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco FXO Hello All, I have a router with an fxo that I would like to tie into Asterisk. I have the sip configuration down and used the example at http://www.voip-info.org/wiki-Asterisk+cisco+FXO to create a config for the Cisco router (3640). Although the router detects the vic, I am unable to set the sipv2 protocol. The only protocol available on this router seems to be "cisco" (I assume that means callmanager). I was wondering if anyone out there knew the type of IOS I should get loaded on this bad boy to support sip. Thanks, Erik ___ 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 ___ 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
[Asterisk-Users] Cisco FXO
Hello All, I have a router with an fxo that I would like to tie into Asterisk. I have the sip configuration down and used the example at http://www.voip-info.org/wiki-Asterisk+cisco+FXO to create a config for the Cisco router (3640). Although the router detects the vic, I am unable to set the sipv2 protocol. The only protocol available on this router seems to be "cisco" (I assume that means callmanager). I was wondering if anyone out there knew the type of IOS I should get loaded on this bad boy to support sip. Thanks, Erik ___ 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
Re: [Asterisk-Users] Cisco FXO
I didn't see an IP Voice, but I did see an IP Plus and an IP/H323. I thought the IP Plus would have it, but was concerned. I know that Asterisk supports h323 so I was considering that one. Anyways thanks for the feedback. Erik On Wed, 13 Oct 2004 21:53:25 -0700, Erik Espinoza <[EMAIL PROTECTED]> wrote: > I didn't see an IP Voice, I did see an IP Plus however... > > > > > On Wed, 13 Oct 2004 20:57:22 -0500, Henry Devito <[EMAIL PROTECTED]> wrote: > > You need to upgrade to the latest IOS. IP voice or IP plus for full SIP > > features. > > > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Erik Espinoza > > Sent: Wednesday, October 13, 2004 8:24 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] Cisco FXO > > > > Hello All, > > > > I have a router with an fxo that I would like to tie into Asterisk. I > > have the sip configuration down and used the example at > > http://www.voip-info.org/wiki-Asterisk+cisco+FXO to create a config > > for the Cisco router (3640). Although the router detects the vic, I am > > unable to set the sipv2 protocol. > > > > The only protocol available on this router seems to be "cisco" (I > > assume that means callmanager). I was wondering if anyone out there > > knew the type of IOS I should get loaded on this bad boy to support > > sip. > > > > Thanks, > > Erik > > ___ > > 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 > > > > > ___ 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
Re: [Asterisk-Users] Cisco FXO as PSTN gateway (updated request for assistance)
Olle E. Johansson wrote: I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my default [bogon-calls] context, not in [pstn-incoming] As I understand it, the Cisco is not registred with Asterisk as a peer. It /appears/ to be: redcusr01*CLI> sip show peers Name/usernameHost Mask Port Status 1001/100110.129.3.128(D) 255.255.255.255 5060 Unmonitored PSTN 10.129.3.254 255.255.255.255 5060 Unmonitored Could you please mail a SIP DEBUG output of an incoming INVITE from the Cisco to Asterisk? redcusr01*CLI> SIP DEBUG SIP Debugging Enabled redcusr01*CLI> Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.129.3.254:5060 From: ;tag=23D83A04-449 To: Date: Sun, 07 Mar 1993 23:02:53 GMT Call-ID: [EMAIL PROTECTED] Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 3055577250-442896844-2154029736-727233534 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: ;party=calling;screen=no;privacy=off Timestamp: 731545373 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 247 v=0 o=CiscoSystemsSIP-GW-UserAgent 7818 6792 IN IP4 10.129.3.254 s=SIP Call c=IN IP4 10.129.3.254 t=0 0 m=audio 19058 RTP/AVP 0 101 c=IN IP4 10.129.3.254 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 20 headers, 11 lines Using latest request as basis request Sending to 10.129.3.254 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format telephone-event Capabilities: us - 4, them - 4/0, combined - 4 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 1001 in bogon-calls list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying All guidance very much welcomed :) Other options I'm considering to fix this are: (1) Using SER to take the incoming calls from the Cisco (2) Using H.323 to take the incoming calls from the Cisco Commetns on these 2 options also welcomed :) Best Wishes, Fran. ___ 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
Re: [Asterisk-Users] Cisco FXO as PSTN gateway (updated request for assistance)
Fran Boon wrote: I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my default [bogon-calls] context, not in [pstn-incoming] Hmmm. As I understand it, the Cisco is not registred with Asterisk as a peer. Could you please mail a SIP DEBUG output of an incoming INVITE from the Cisco to Asterisk? /O ___ 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
[Asterisk-Users] Cisco FXO as PSTN gateway (updated request for assistance)
I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my default [bogon-calls] context, not in [pstn-incoming] Can anyone help me locate why? (Config files are on the Wiki) I have done a packet sniff & decoded using Ethereal-0.10.0, but this doesn't tell me a great deal - I just see the rejection message: y.y.y.y x.x.x.x INVITE sip:[EMAIL PROTECTED]:5060 x.x.x.x y.y.y.y Status: 100 Trying x.x.x.x y.y.y.y Status: 503 Service Unavailable y.y.y.y x.x.x.x Request: ACK sip:[EMAIL PROTECTED]:5060 (resent as retested with 0.7.1 & the addition of autocreatepeer=yes) Thanks a lot, Fran. ___ 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
Re: [Asterisk-Users] Cisco FXO as PSTN gateway
Philipp von Klitzing wrote: I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO Try this in your [pstn] section: autocreatepeer=yes Thanks for the suggestion - hasn't helped though :/ 'sip show peers' looks the same in each case. Any more ideas? Add this to each (!) of your SIP users: disallow=all allow=ulaw Is having this in the [general] section not enough? My endpoints are all using ulaw anyway & surely this won't affect which context the call comes in as? Also: The syntax is "username=" and not "user=". Same for "dtmfmode" instead of "dtmf". Well-spotted - I was actually ok in the real machine - it was a bad copy to the Wiki (now corrected) F ___ 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
Re: [Asterisk-Users] Cisco FXO as PSTN gateway
Philipp von Klitzing wrote: Hi! I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO Try this in your [pstn] section: autocreatepeer=yes Add this to each (!) of your SIP users: disallow=all allow=ulaw Also: The syntax is "username=" and not "user=". Same for "dtmfmode" instead of "dtmf". ...and if you want to live dangerously, try out the chan_sip2 in bugs.digium.com You can add more options to autocreated peers with the [template] functionality in that version of the sip channel. But remember it's a beta and I need user feedback. /O :-) ___ 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
Re: [Asterisk-Users] Cisco FXO as PSTN gateway
Hi! > I have been compiling information on this configuration onto the Wiki: > http://voip-info.org/wiki-Asterisk+cisco+FXO Try this in your [pstn] section: autocreatepeer=yes Add this to each (!) of your SIP users: disallow=all allow=ulaw Also: The syntax is "username=" and not "user=". Same for "dtmfmode" instead of "dtmf". Cheers, Philipp ___ 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
[Asterisk-Users] Cisco FXO as PSTN gateway
I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my [bogon-calls] context. Can anyone help me locate why? (Config files are on the Wiki) I have done a packet sniff & decoded using Ethereal-0.10.0, but this doesn't tell me a great deal - I just see the rejection message: y.y.y.y x.x.x.x INVITE sip:[EMAIL PROTECTED]:5060 x.x.x.x y.y.y.y Status: 100 Trying x.x.x.x y.y.y.y Status: 503 Service Unavailable y.y.y.y x.x.x.x Request: ACK sip:[EMAIL PROTECTED]:5060 Thanks a lot, Fran. ___ 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
