Mark Dutton
2005-May-22 05:55 UTC
[Asterisk-Users] Getting a Cisco gateway to work with Asterisk
Can anyone please help me with sample IOS commands to get a Cisco gateway working properly with Asterisk. I cannot get my Cisco 2801 with BRI interfaces to call into Asterisk. The Cisco identifies itself as sip:.@datamerge.local. I cannot figure out how to get it to identify as sip:cisco@datamerge.local. The gateway works with other SIP servers that don't require authentication, but Asterisk wants it to authenticate, or at least idenitify itself and I cannot work this bit out. If I put in the host address in my sip.conf, I still get a "cannot find host 192.168.44.23:<random port number>, where <random port number> is actually some random port number. I am at my wits end. Regards Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050522/bc30786e/attachment.htm
Steve Blair
2005-May-22 08:10 UTC
[Asterisk-Users] Getting a Cisco gateway to work with Asterisk
When you say identify I presume you are trying to get the Cisco to register as a user. To the best of my knowledge it cannot do this. Instead define a peer in sip.conf which is the gateway and place traffic matching this peer into a context that is defined in your extensions.conf file. The Cisco will need dial-peer statements to match inbound dialed digits and forward all matching calls to your Asterisk box. Mark Dutton wrote:> Can anyone please help me with sample IOS commands to get a Cisco > gateway working properly with Asterisk. > > I cannot get my Cisco 2801 with BRI interfaces to call into Asterisk. > > The Cisco identifies itself as sip:.@datamerge.local. > > I cannot figure out how to get it to identify as > sip:cisco@datamerge.local. The gateway works with other SIP servers > that don't require authentication, but Asterisk wants it to > authenticate, or at least idenitify itself and I cannot work this bit out. > > If I put in the host address in my sip.conf, I still get a "cannot > find host 192.168.44.23:<random port number>, where <random port > number> is actually some random port number. > > I am at my wits end. > > Regards > > Mark > >------------------------------------------------------------------------ > >_______________________________________________ >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 >
barney
2005-May-23 00:39 UTC
[Asterisk-Users] Getting a Cisco gateway to work with Asterisk
I have that same problem just now. I`m trying to find some solution with
serveral tests, using IOS v.12.3(8r)T7 on the C2821 box with two PRI ports.
When i find something, it`ll be posted here, and i`m awaiting to do it also from
your side.
-b
----- Original Message -----
From: Mark Dutton
To: asterisk-users@lists.digium.com
Sent: Sunday, May 22, 2005 2:55 PM
Subject: [Asterisk-Users] Getting a Cisco gateway to work with Asterisk
Can anyone please help me with sample IOS commands to get a Cisco gateway
working properly with Asterisk.
I cannot get my Cisco 2801 with BRI interfaces to call into Asterisk.
The Cisco identifies itself as sip:.@datamerge.local.
I cannot figure out how to get it to identify as sip:cisco@datamerge.local.
The gateway works with other SIP servers that don't require authentication,
but Asterisk wants it to authenticate, or at least idenitify itself and I cannot
work this bit out.
If I put in the host address in my sip.conf, I still get a "cannot find
host 192.168.44.23:<random port number>, where <random port number>
is actually some random port number.
I am at my wits end.
Regards
Mark
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Mark Dutton
2005-May-23 06:02 UTC
[Asterisk-Users] Getting a Cisco gateway to work with Asterisk
Thanks Steve I realised the other day that I don't want the Cisco to register with credentials. There is in fact a hidden credentials command in 12.3(8)T. What I did was take away all registration commands from my sip-ua block in the Cisco. I am using asterisk@home, so I have created a trunk through AMP. I have changed the settings in outbound trunk to the following and created an empty inbound trunk on the web page with no parameters. The result is that in Asterisk sip_additional.conf I have this block [cisco] context=from-pstn host=192.168.44.23 type=friend Now when I try to call into my gateway from the PSTN, I get the following line immediately after the Cisco does an invite Sip read: INVITE sip:390@dev.datamerge.local:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.23:5060;branch=z9hG4bK3016D6 From: <sip:894742460@dev.datamerge.local>;tag=391004-1A5E To: <sip:390@dev.datamerge.local> Date: Sun, 22 May 2005 14:29:25 GMT Call-ID: BB5B196D-CA0411D9-803BE53F-D6B5D89@192.168.44.23 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 3143229573-3389264345-2148466707-2141291050 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: 15 Remote-Party-ID: <sip:894742460@192.168.44.23>;party=calling;screen=yes;privacy=off Timestamp: 1116772165 Contact: <sip:894742460@192.168.44.23:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 328 v=0 o=CiscoSystemsSIP-GW-UserAgent 23 4058 IN IP4 192.168.44.23 s=SIP Call c=IN IP4 192.168.44.23 t=0 0 m=audio 17780 RTP/AVP 8 18 98 3 0 19 c=IN IP4 192.168.44.23 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:98 GSM-EFR/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:19 CN/8000 20 headers, 14 lines Using latest request as basis request Sending to 192.168.44.23 : 5060 (non-NAT) Found no matching peer or user for '192.168.44.23:57704' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 98 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 19 Peer audio RTP is at port 192.168.44.23:17780 Found description format PCMA Found description format G729 Found description format GSM-EFR Found description format GSM Found description format PCMU Found description format CN Capabilities: us - 0x508 (alaw|g729|ilbc), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x2 (gsm), combined - 0x0 (nothing) Looking for 390 in from-sip-external list_route: hop: <sip:894742460@192.168.44.23:5060> You can see the line Found no matching peer or user for '192.168.44.23:57704' OK, now if I go into the parameters for my trunk and add the line Port=57704 It works!!! Problem is, the port changes. The question then is, where in my Cisco config can I specify the listening (or return) port to 5060 so it does not pick an arbitrary port from the pool? Regards Mark Date: Sun, 22 May 2005 11:10:31 -0400 From: Steve Blair <blairs@isc.upenn.edu> Subject: Re: [Asterisk-Users] Getting a Cisco gateway to work with Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <4290A0E7.9010505@isc.upenn.edu> Content-Type: text/plain; charset=ISO-8859-1; format=flowed When you say identify I presume you are trying to get the Cisco to register as a user. To the best of my knowledge it cannot do this. Instead define a peer in sip.conf which is the gateway and place traffic matching this peer into a context that is defined in your extensions.conf file. The Cisco will need dial-peer statements to match inbound dialed digits and forward all matching calls to your Asterisk box. Mark Dutton wrote:> Can anyone please help me with sample IOS commands to get a Cisco > gateway working properly with Asterisk. > > I cannot get my Cisco 2801 with BRI interfaces to call into Asterisk. > > The Cisco identifies itself as sip:.@datamerge.local. > > I cannot figure out how to get it to identify as > sip:cisco@datamerge.local. The gateway works with other SIP servers > that don't require authentication, but Asterisk wants it to > authenticate, or at least idenitify itself and I cannot work this bit out. > > If I put in the host address in my sip.conf, I still get a "cannot > find host 192.168.44.23:<random port number>, where <random port > number> is actually some random port number. > > I am at my wits end. > > Regards > > Mark > >----------------------------------------------------------------------- >-
niels@wxn.nl
2005-May-23 08:47 UTC
[Asterisk-Users] Getting a Cisco gateway to work with Asterisk
Try setting defaultip=192.168.44.23 Too -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of barney Sent: Monday, May 23, 2005 4:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Getting a Cisco gateway to work with Asterisk I tried that, but it is not working for me with Asterisk@Home v1.0 :-( -b> Mark, > > Try writing the sip.conf stanza as: > > [192.168.44.23] > context=from-pstn > host=192.168.44.23 > type=friend > insecure=very > > The 'insecure=very' allows any calls from this IP address to match. > > Alistair Cunningham, > Integrics Ltd, > +44 (0)7870 699 479 > http://integrics.com/ > > > Mark Dutton wrote: >> Thanks Steve >> >> I realised the other day that I don't want the Cisco to register with>> credentials. There is in fact a hidden credentials command in12.3(8)T.>> >> What I did was take away all registration commands from my sip-ua >> block in the Cisco. >> >> I am using asterisk@home, so I have created a trunk through AMP. I >> have changed the settings in outbound trunk to the following and >> created an empty inbound trunk on the web page with no parameters. >> >> The result is that in Asterisk sip_additional.conf I have this block >> >> [cisco] >> context=from-pstn >> host=192.168.44.23 >> type=friend >> >> Now when I try to call into my gateway from the PSTN, I get the >> following line immediately after the Cisco does an invite >> >> Sip read: INVITE sip:390@dev.datamerge.local:5060 SIP/2.0 Via: >> SIP/2.0/UDP 192.168.44.23:5060;branch=z9hG4bK3016D6 From: >> <sip:894742460@dev.datamerge.local>;tag=391004-1A5E To: >> <sip:390@dev.datamerge.local> Date: Sun, 22 May 2005 14:29:25 GMT >> Call-ID: BB5B196D-CA0411D9-803BE53F-D6B5D89@192.168.44.23 Supported: >> 100rel,timer Min-SE: 1800 Cisco-Guid: >> 3143229573-3389264345-2148466707-2141291050 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: 15 >> Remote-Party-ID: >> <sip:894742460@192.168.44.23>;party=calling;screen=yes;privacy=off >> Timestamp: 1116772165 Contact: <sip:894742460@192.168.44.23:5060> >> Expires: 180 Allow-Events: telephone-event Content-Type: >> application/sdp >> Content-Length: 328 v=0 o=CiscoSystemsSIP-GW-UserAgent 23 4058 IN >> IP4 >> 192.168.44.23 s=SIP Call c=IN IP4 192.168.44.23 t=0 0 m=audio 17780 >> RTP/AVP 8 18 98 3 0 19 c=IN IP4 192.168.44.23 a=rtpmap:8 PCMA/8000 >> a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:98 GSM-EFR/8000 >> a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:19 CN/8000 20 >> headers, >> 14 lines >> Using latest request as basis request Sending to 192.168.44.23 : >> 5060 (non-NAT) Found no matching peer or user for >> '192.168.44.23:57704' >> Found RTP audio format 8 >> Found RTP audio format 18 >> Found RTP audio format 98 >> Found RTP audio format 3 >> Found RTP audio format 0 >> Found RTP audio format 19 >> Peer audio RTP is at port 192.168.44.23:17780 Found description >> format PCMA Found description format G729 Found description format >> GSM-EFR Found description format GSM Found description format PCMU>> Found description format CN >> Capabilities: us - 0x508 (alaw|g729|ilbc), peer - audio=0x10e >> (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x108 >> (alaw|g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x2 >> (gsm), combined - 0x0 >> (nothing) >> Looking for 390 in from-sip-external >> list_route: hop: <sip:894742460@192.168.44.23:5060> >> >> You can see the line Found no matching peer or user for >> '192.168.44.23:57704' >> >> OK, now if I go into the parameters for my trunk and add the line >> >> Port=57704 >> >> It works!!! >> >> Problem is, the port changes. The question then is, where in my Cisco>> config can I specify the listening (or return) port to 5060 so it >> does not pick an arbitrary port from the pool? >> >> Regards >> >> Mark >> >> >> >> Date: Sun, 22 May 2005 11:10:31 -0400 >> From: Steve Blair <blairs@isc.upenn.edu> >> Subject: Re: [Asterisk-Users] Getting a Cisco gateway to work with >> Asterisk >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <asterisk-users@lists.digium.com> >> Message-ID: <4290A0E7.9010505@isc.upenn.edu> >> Content-Type: text/plain; charset=ISO-8859-1; format=flowed >> >> >> When you say identify I presume you are trying to get the Cisco to >> register as a user. To the best of my knowledge it cannot do this. >> Instead >> define a peer in sip.conf which is the gateway and place traffic >> matching this peer into a context that is defined in yourextensions.conf file.>> The >> Cisco will need dial-peer statements to match inbound dialed digits >> and forward all matching calls to your Asterisk box. >> >> >> >> Mark Dutton wrote: >> >> >>>Can anyone please help me with sample IOS commands to get a Cisco >>>gateway working properly with Asterisk. >>> I cannot get my Cisco 2801 with BRI interfaces to call intoAsterisk.>>> The Cisco identifies itself as sip:.@datamerge.local. >>> I cannot figure out how to get it to identify as >>>sip:cisco@datamerge.local. The gateway works with other SIP servers >>>that don't require authentication, but Asterisk wants it to >>>authenticate, or at least idenitify itself and I cannot work thisbit out.>>> If I put in the host address in my sip.conf, I still get a "cannot >>>find host 192.168.44.23:<random port number>, where <random port >>>number> is actually some random port number. >>> I am at my wits end. >>> Regards >>> Mark >>> >>>--------------------------------------------------------------------- >>>-- >>>- >> >> >> _______________________________________________ >> 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 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 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