RE: [Asterisk-Users] Getting a Cisco gateway to work with Asterisk
Try setting defaultip=192.168.44.23 Too -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 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 [EMAIL PROTECTED], 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:[EMAIL PROTECTED]:5060 SIP/2.0 Via: >> SIP/2.0/UDP 192.168.44.23:5060;branch=z9hG4bK3016D6 From: >> ;tag=391004-1A5E To: >> Date: Sun, 22 May 2005 14:29:25 GMT >> Call-ID: [EMAIL PROTECTED] 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: >> ;party=calling;screen=yes;privacy=off >> Timestamp: 1116772165 Contact: >> 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: >> >> 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 <[EMAIL PROTECTED]> >> Subject: Re: [Asterisk-Users] Getting a Cisco gateway to work with >> Asterisk >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> >> Message-ID: <[EMAIL PROTECTED]> >> 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 knowl
Re: [Asterisk-Users] Getting a Cisco gateway to work with Asterisk
I tried that, but it is not working for me with [EMAIL PROTECTED] 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 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 [EMAIL PROTECTED], 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:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.23:5060;branch=z9hG4bK3016D6 From: ;tag=391004-1A5E To: Date: Sun, 22 May 2005 14:29:25 GMT Call-ID: [EMAIL PROTECTED] 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: ;party=calling;screen=yes;privacy=off Timestamp: 1116772165 Contact: 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: 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 <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Getting a Cisco gateway to work with Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] I cannot figure out how to get it to identify as sip:[EMAIL PROTECTED] 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:, where 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
Re: [Asterisk-Users] Getting a Cisco gateway to work with Asterisk
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 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 [EMAIL PROTECTED], 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:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.23:5060;branch=z9hG4bK3016D6 From: ;tag=391004-1A5E To: Date: Sun, 22 May 2005 14:29:25 GMT Call-ID: [EMAIL PROTECTED] 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: ;party=calling;screen=yes;privacy=off Timestamp: 1116772165 Contact: 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: 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 <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Getting a Cisco gateway to work with Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] I cannot figure out how to get it to identify as sip:[EMAIL PROTECTED] 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:, where 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
Re: [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 [EMAIL PROTECTED], 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:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.23:5060;branch=z9hG4bK3016D6 From: ;tag=391004-1A5E To: Date: Sun, 22 May 2005 14:29:25 GMT Call-ID: [EMAIL PROTECTED] 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: ;party=calling;screen=yes;privacy=off Timestamp: 1116772165 Contact: 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: 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 <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Getting a Cisco gateway to work with Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] > > I cannot figure out how to get it to identify as > sip:[EMAIL PROTECTED] 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:, where 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
Re: [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:[EMAIL PROTECTED] I cannot figure out how to get it to identify as sip:[EMAIL PROTECTED] 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:, where is actually some random port number. I am at my wits end. Regards Mark ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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
Re: [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:[EMAIL PROTECTED] I cannot figure out how to get it to identify as sip:[EMAIL PROTECTED] 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:, where 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] 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:[EMAIL PROTECTED] I cannot figure out how to get it to identify as sip:[EMAIL PROTECTED] 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:, where 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
