Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"
After many hours of fiddling around, Andres gave me the final piece. For those looking to implement SIP Trunks on a CS-1500 with Asterisk, here are the pieces: Diagram: CS-1500 -- customer PBX (172.16.10.40)(172.16.10.195) HOST: should be the DNS name assigned to the CS-1500's SIP interface. e.g. sip.acme.com NUSR: user name used for the CS 1500 to login into the customer PBX. Needs to match up FreePBX's "Trunk Name". For those who use the CLI, this section in sip.conf is encased in square brackets. i.e. [customername] NPSW: password used for the CS 1500 to login into the customer PBX. Needs to match up with the secret= line. i.e. secret=password IP: IP address of the customer PBX. i.e. 172.16.10.195 LUSR: user name used for the customer PBX to login into the CS 1500. Needs to match up with the username= line. i.e. username=customername LPSW: password used for the customer PBX to login into the CS 1500. Needs to match up with the secret= line. i.e. secret=password. For simplicity we made NUSR/LUSR the same and NPSW/LPSW the same. Since you need to define a trunk per customer, it makes the most sense and it easiest to support and implement. Here's what you need to add to Asterisk's sip.conf (yes, just those few lines!) [customername] host=sip.acme.com type=friend username=customername secret=password And the CS-1500 output: TYP TG NUM 1234 TGTP 2WAY TGNM SIP MG NO SIGT SIP STSI 0 HNPA 555 RC 0 RTP 0 TRNL PRFX PRFX 24 APFX NONE TRFC NONE 4XCD YES ACKA NO TYPC NOCO NXX UNKN LATA 000 CMCT NO TGID NONE SIT NO CNAR NO LRN NONE TNDM NO LDAT NO TRFC NONE EOAT NO ATIC NO CMCO NO TGMU NO HOST sip.acme.com NUSR customername NPSW password IP 172.16.10.195 PORT 5060 PROT UDP T38F NO AUTH YES LUSR customername LPSW password CLIM 7 CPBY 0 Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Bulk - iName.com Sent: Monday, January 05, 2009 6:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very" The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add "insecure=very" to my "Outgoing settings", but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with "SIP/2.0 403 Forbidden" I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the "Incoming Settings" panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com SIP/2.0 From: ;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2db ba4 To: Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: Privacy: none Remote-Party-ID: ; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: Authorization: Digest username="username",realm="asterisk",nonce="118af2b0",uri="sip:+15552027020@ sip.acme.com",response="111e63ec2a1f3ebabefe4f7dae4087a1",algorithm=MD5 Content-Type: application/SDP Content-Length: 167 v=0 o=- 2973921782 2973921782 IN IP4 172.16.10.65 s=SIP Call c=IN IP4 172.16.10.65 t=0 0 m=audio 36224 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"
You're the miracle worker! Thanks! Frank From: Andres [mailto:and...@telesip.net] Sent: Tuesday, January 06, 2009 11:19 AM To: Frank Bulk Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very" Frank Bulk wrote: This is what I have in my configuration now: [ACME] host=sip.acme.com username=username secret=password type=friend Your problem is you are trying to do authenticate by host and by username at the same time. That does not work in asterisk. You should be seeing a Warning message in the console saying something like: check_auth: username mismatch, have , digest has That means you already matched to sip.conf entry ACME, but the digest has a different username, so it fails. You can fix it by setting the paramters in the CS1500 to have the username = ACME. That way the digest will come in as: Digest username="ACME" ...bla bla bla Andres http://www.telesip.net I've done a SIP debug before, but I've done it again with the above configuration: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 after which "SIP/2.0 401 Unauthorized" is issued after the un-authenticated INVITE and "SIP/2.0 403 Forbidden" after the authenticated INVITE. When I add "insecure=very", this is what the SIP debug shows: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 Found RTP audio format 0 Peer audio RTP is at port 172.16.10.65:36272 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 172.16.10.65:36272 Looking for +15552127020 in from-sip-external (domain sip.acme.com) list_route: hop: It isn't very clear (to me) from the success how the "insecure=very" helps. Frank -Original Message- From: Andres [mailto:and...@telesip.net] Sent: Monday, January 05, 2009 7:43 PM To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very" Frank Bulk - iName.com wrote: The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add "insecure=very" to my "Outgoing settings", but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with "SIP/2.0 403 Forbidden" I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the "Incoming Settings" panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Do a sip debug on the asterisk console and see if it is actually is matching one of your sip.conf entries during an invite from the CS1500. Look for a line that says something like 'Found Peerbla bla bla'. If you dont see that line, then you are not even adding the correct sip.conf entry to match the invite from the CS1500. Andres http://www.telesip.net Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com SIP/2.0 From: ;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d b ba4 To: Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: Privacy: none Remote-Party-ID: ; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: Authorization: Digest username="username",realm="asterisk",nonce="118af2b0",uri="sip:+15552027020 @ sip.acme.com",response="111e63ec2a1f3ebabefe4f7dae4087a1",algorithm=MD5 Content-Type: application/SDP Content-Length: 167 v=0 o=- 2973921782 2973921782 IN IP4 172.16.10.65 s=SIP Call c=IN IP4 172.16.10.65 t=0 0 m=au
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"
Frank Bulk wrote: This is what I have in my configuration now: [ACME] host=sip.acme.com username=username secret=password type=friend Your problem is you are trying to do authenticate by host and by username at the same time. That does not work in asterisk. You should be seeing a Warning message in the console saying something like: check_auth: username mismatch, have , digest has That means you already matched to sip.conf entry ACME, but the digest has a different username, so it fails. You can fix it by setting the paramters in the CS1500 to have the username = ACME. That way the digest will come in as: Digest username="ACME" ...bla bla bla Andres http://www.telesip.net I've done a SIP debug before, but I've done it again with the above configuration: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 after which "SIP/2.0 401 Unauthorized" is issued after the un-authenticated INVITE and "SIP/2.0 403 Forbidden" after the authenticated INVITE. When I add "insecure=very", this is what the SIP debug shows: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 Found RTP audio format 0 Peer audio RTP is at port 172.16.10.65:36272 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 172.16.10.65:36272 Looking for +15552127020 in from-sip-external (domain sip.acme.com) list_route: hop: It isn't very clear (to me) from the success how the "insecure=very" helps. Frank -Original Message- From: Andres [mailto:and...@telesip.net] Sent: Monday, January 05, 2009 7:43 PM To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very" Frank Bulk - iName.com wrote: The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add "insecure=very" to my "Outgoing settings", but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with "SIP/2.0 403 Forbidden" I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the "Incoming Settings" panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Do a sip debug on the asterisk console and see if it is actually is matching one of your sip.conf entries during an invite from the CS1500. Look for a line that says something like 'Found Peerbla bla bla'. If you dont see that line, then you are not even adding the correct sip.conf entry to match the invite from the CS1500. Andres http://www.telesip.net Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com SIP/2.0 From: ;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d b ba4 To: Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: Privacy: none Remote-Party-ID: ; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: Authorization: Digest username="username",realm="asterisk",nonce="118af2b0",uri="sip:+15552027020 @ sip.acme.com",response="111e63ec2a1f3ebabefe4f7dae4087a1",algorithm=MD5 Content-Type: application/SDP Content-Length: 167 v=0 o=- 2973921782 2973921782 IN IP4 172.16.10.65 s=SIP Call c=IN IP4 172.16.10.65 t=0 0 m=audio 36224 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"
That's a good suggestion, but I tried that and it didn't work. I think you need an '&' in-between, so I tried that, too. I also tried adding the IP address of the CS 1500, too, and that didn't help. Frank From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grygoriy Dobrovolskyy Sent: Tuesday, January 06, 2009 2:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very" try do add fromdomain=acme.com/sip.acme.com fromhost=acme.com/sip.acme.com 2009/1/6 Frank Bulk I tried that before, but I just tried it again. Unfortunately, the same thing: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 [ACME] host=172.16.10.40 username=username secret=password type=friend Frank From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Allan Dib Sent: Monday, January 05, 2009 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very" Try it by IP address instead of hostname as reverse DNS may not be resolving. e.g. host=123.123.123.123 On Tue, Jan 6, 2009 at 2:25 PM, Frank Bulk wrote: This is what I have in my configuration now: [ACME] host=sip.acme.com username=username secret=password type=friend I've done a SIP debug before, but I've done it again with the above configuration: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 after which "SIP/2.0 401 Unauthorized" is issued after the un-authenticated INVITE and "SIP/2.0 403 Forbidden" after the authenticated INVITE. When I add "insecure=very", this is what the SIP debug shows: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 Found RTP audio format 0 Peer audio RTP is at port 172.16.10.65:36272 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 172.16.10.65:36272 Looking for +15552127020 in from-sip-external (domain sip.acme.com) list_route: hop: mailto:sip%3a5551236...@172.16.10.40> > It isn't very clear (to me) from the success how the "insecure=very" helps. Frank -Original Message- From: Andres [mailto:and...@telesip.net] Sent: Monday, January 05, 2009 7:43 PM To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very" Frank Bulk - iName.com wrote: >The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not >work unless I add "insecure=very" to my "Outgoing settings", but I don't >want to do that. I do want to authenticate. Outgoing (Asterisk PBX to >Class 5 switch) calls do authenticate and work. > >The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username >and password that it's sending out. But the INVITE is responded by the >Asterisk with "SIP/2.0 403 Forbidden" > >I've changed the INVITE message to mask the real telephone numbers, SIP >server, passwords, and IP addresses, but I did that using search and replace >so the structure is intact. > >What do I need to configure in the "Incoming Settings" panel for the CS >1500's INVITE to my Asterisk server to work? I've tried all kinds of >combinations of user,username,authname using +15552027020,host with IP >and/or DNS name, but nothing appears to work. > > > Do a sip debug on the asterisk console and see if it is actually is matching one of your sip.conf entries during an invite from the CS1500. Look for a line that says something like 'Found Peerbla bla bla'. If you dont see that line, then you are not even adding the correct sip.conf entry to match the invite from the CS1500. Andres http://www.telesip.net >Frank > >INVITE message from Wireshark packet capture: > >INVITE sip:+15552027...@sip.acme.com <mailto:sip%3a%2b15552027...@sip.acme.com> SIP/2.0 >From: >mailto:sip%3a5552022...@172.16.10.40> >;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d b >ba4 >To: mailto:sip%3a%2b15552027...@sip.acme.com> > >Call-ID: f379f6
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"
try do add fromdomain=acme.com/sip.acme.com fromhost=acme.com/sip.acme.com 2009/1/6 Frank Bulk > I tried that before, but I just tried it again. Unfortunately, the same > thing: > > No user '5551236049' in SIP users list > > Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 > > > > [ACME] > host=172.16.10.40 > username=username > secret=password > type=friend > > > > Frank > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Allan Dib > *Sent:* Monday, January 05, 2009 9:41 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* Re: [asterisk-users] Incoming side of SIP trunk does not work > unless I add "insecure=very" > > > > Try it by IP address instead of hostname as reverse DNS may not be > resolving. e.g. host=123.123.123.123 > > On Tue, Jan 6, 2009 at 2:25 PM, Frank Bulk wrote: > > This is what I have in my configuration now: > > [ACME] > host=sip.acme.com > username=username > secret=password > type=friend > > I've done a SIP debug before, but I've done it again with the above > configuration: >No user '5551236049' in SIP users list >Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 > after which "SIP/2.0 401 Unauthorized" is issued after the un-authenticated > INVITE and "SIP/2.0 403 Forbidden" after the authenticated INVITE. > > When I add "insecure=very", this is what the SIP debug shows: >No user '5551236049' in SIP users list >Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 >Found RTP audio format 0 >Peer audio RTP is at port 172.16.10.65:36272 >Found audio description format PCMU for ID 0 >Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 > (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) >Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - > 0x0 (nothing), combined - 0x0 (nothing) >Peer audio RTP is at port 172.16.10.65:36272 >Looking for +15552127020 in from-sip-external (domain sip.acme.com) >list_route: hop: > > > > > It isn't very clear (to me) from the success how the "insecure=very" helps. > > Frank > > > -Original Message- > From: Andres [mailto:and...@telesip.net] > Sent: Monday, January 05, 2009 7:43 PM > To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial > Discussion > > Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work > unless I add "insecure=very" > > Frank Bulk - iName.com wrote: > > >The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not > >work unless I add "insecure=very" to my "Outgoing settings", but I don't > >want to do that. I do want to authenticate. Outgoing (Asterisk PBX to > >Class 5 switch) calls do authenticate and work. > > > >The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a > username > >and password that it's sending out. But the INVITE is responded by the > >Asterisk with "SIP/2.0 403 Forbidden" > > > >I've changed the INVITE message to mask the real telephone numbers, SIP > >server, passwords, and IP addresses, but I did that using search and > replace > >so the structure is intact. > > > >What do I need to configure in the "Incoming Settings" panel for the CS > >1500's INVITE to my Asterisk server to work? I've tried all kinds of > >combinations of user,username,authname using +15552027020,host with IP > >and/or DNS name, but nothing appears to work. > > > > > > > Do a sip debug on the asterisk console and see if it is actually is > matching one of your sip.conf entries during an invite from the CS1500. > Look for a line that says something like 'Found Peerbla bla bla'. > If you dont see that line, then you are not even adding the correct > sip.conf entry to match the invite from the CS1500. > > Andres > http://www.telesip.net > > >Frank > > > >INVITE message from Wireshark packet capture: > > > >INVITE sip:+15552027...@sip.acme.com > >SIP/2.0 > >From: > > > >;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d > b > >ba4 > >To: > > >Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 > >CSeq: 5102 INVITE > >Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7c
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"
I tried that before, but I just tried it again. Unfortunately, the same thing: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 [ACME] host=172.16.10.40 username=username secret=password type=friend Frank From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Allan Dib Sent: Monday, January 05, 2009 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very" Try it by IP address instead of hostname as reverse DNS may not be resolving. e.g. host=123.123.123.123 On Tue, Jan 6, 2009 at 2:25 PM, Frank Bulk wrote: This is what I have in my configuration now: [ACME] host=sip.acme.com username=username secret=password type=friend I've done a SIP debug before, but I've done it again with the above configuration: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 after which "SIP/2.0 401 Unauthorized" is issued after the un-authenticated INVITE and "SIP/2.0 403 Forbidden" after the authenticated INVITE. When I add "insecure=very", this is what the SIP debug shows: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 Found RTP audio format 0 Peer audio RTP is at port 172.16.10.65:36272 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 172.16.10.65:36272 Looking for +15552127020 in from-sip-external (domain sip.acme.com) list_route: hop: mailto:sip%3a5551236...@172.16.10.40> > It isn't very clear (to me) from the success how the "insecure=very" helps. Frank -Original Message- From: Andres [mailto:and...@telesip.net] Sent: Monday, January 05, 2009 7:43 PM To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very" Frank Bulk - iName.com wrote: >The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not >work unless I add "insecure=very" to my "Outgoing settings", but I don't >want to do that. I do want to authenticate. Outgoing (Asterisk PBX to >Class 5 switch) calls do authenticate and work. > >The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username >and password that it's sending out. But the INVITE is responded by the >Asterisk with "SIP/2.0 403 Forbidden" > >I've changed the INVITE message to mask the real telephone numbers, SIP >server, passwords, and IP addresses, but I did that using search and replace >so the structure is intact. > >What do I need to configure in the "Incoming Settings" panel for the CS >1500's INVITE to my Asterisk server to work? I've tried all kinds of >combinations of user,username,authname using +15552027020,host with IP >and/or DNS name, but nothing appears to work. > > > Do a sip debug on the asterisk console and see if it is actually is matching one of your sip.conf entries during an invite from the CS1500. Look for a line that says something like 'Found Peerbla bla bla'. If you dont see that line, then you are not even adding the correct sip.conf entry to match the invite from the CS1500. Andres http://www.telesip.net >Frank > >INVITE message from Wireshark packet capture: > >INVITE sip:+15552027...@sip.acme.com <mailto:sip%3a%2b15552027...@sip.acme.com> SIP/2.0 >From: >mailto:sip%3a5552022...@172.16.10.40> >;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d b >ba4 >To: mailto:sip%3a%2b15552027...@sip.acme.com> > >Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 >CSeq: 5102 INVITE >Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 >User-Agent: Nortel CS1500UA/v02.00.REL01 >Accept: application/sdp >P-Asserted-Identity: mailto:sip%3a5552022...@172.16.10.40> ;user=phone> >Privacy: none >Remote-Party-ID: mailto:sip%3a5552022...@172.16.10.40> ;user=phone>; party=calling; >privacy=off >Max-Forwards: 70 >Supported: 100rel,replaces >Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK >Contact: mailto:sip%3a5552022...@172.16.10.40> > >Authorization: Digest >username="username",realm="asterisk",no
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"
Try it by IP address instead of hostname as reverse DNS may not be resolving. e.g. host=123.123.123.123 On Tue, Jan 6, 2009 at 2:25 PM, Frank Bulk wrote: > This is what I have in my configuration now: > > [ACME] > host=sip.acme.com > username=username > secret=password > type=friend > > I've done a SIP debug before, but I've done it again with the above > configuration: >No user '5551236049' in SIP users list >Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 > after which "SIP/2.0 401 Unauthorized" is issued after the un-authenticated > INVITE and "SIP/2.0 403 Forbidden" after the authenticated INVITE. > > When I add "insecure=very", this is what the SIP debug shows: >No user '5551236049' in SIP users list >Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 >Found RTP audio format 0 >Peer audio RTP is at port 172.16.10.65:36272 >Found audio description format PCMU for ID 0 >Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 > (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) >Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - > 0x0 (nothing), combined - 0x0 (nothing) >Peer audio RTP is at port 172.16.10.65:36272 >Looking for +15552127020 in from-sip-external (domain sip.acme.com) >list_route: hop: > > > > > It isn't very clear (to me) from the success how the "insecure=very" helps. > > Frank > > -Original Message- > From: Andres [mailto:and...@telesip.net] > Sent: Monday, January 05, 2009 7:43 PM > To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work > unless I add "insecure=very" > > Frank Bulk - iName.com wrote: > > >The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not > >work unless I add "insecure=very" to my "Outgoing settings", but I don't > >want to do that. I do want to authenticate. Outgoing (Asterisk PBX to > >Class 5 switch) calls do authenticate and work. > > > >The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a > username > >and password that it's sending out. But the INVITE is responded by the > >Asterisk with "SIP/2.0 403 Forbidden" > > > >I've changed the INVITE message to mask the real telephone numbers, SIP > >server, passwords, and IP addresses, but I did that using search and > replace > >so the structure is intact. > > > >What do I need to configure in the "Incoming Settings" panel for the CS > >1500's INVITE to my Asterisk server to work? I've tried all kinds of > >combinations of user,username,authname using +15552027020,host with IP > >and/or DNS name, but nothing appears to work. > > > > > > > Do a sip debug on the asterisk console and see if it is actually is > matching one of your sip.conf entries during an invite from the CS1500. > Look for a line that says something like 'Found Peerbla bla bla'. > If you dont see that line, then you are not even adding the correct > sip.conf entry to match the invite from the CS1500. > > Andres > http://www.telesip.net > > >Frank > > > >INVITE message from Wireshark packet capture: > > > >INVITE sip:+15552027...@sip.acme.com > >SIP/2.0 > >From: > > > >;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d > b > >ba4 > >To: > > >Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 > >CSeq: 5102 INVITE > >Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 > >User-Agent: Nortel CS1500UA/v02.00.REL01 > >Accept: application/sdp > >P-Asserted-Identity: > > > ;user=phone> > >Privacy: none > >Remote-Party-ID: > >;user=phone>; > party=calling; > >privacy=off > >Max-Forwards: 70 > >Supported: 100rel,replaces > >Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK > >Contact: > > >Authorization: Digest > > >username="username",realm="asterisk",nonce="118af2b0",uri="sip:+15552027020 > @ > >sip.acme.com",response="111e63ec2a1f3ebabefe4f7dae4087a1",algorithm=MD5 > >Content-Type: application/SDP > >Content-Length: 167 > > > >v=0 > >o=- 2973921782 2973921782 IN IP4 172.16.10.65 > >s=SIP Call > >c=IN IP4 172.16.10.65 > >t=0 0 > >m=audio 36
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"
This is what I have in my configuration now: [ACME] host=sip.acme.com username=username secret=password type=friend I've done a SIP debug before, but I've done it again with the above configuration: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 after which "SIP/2.0 401 Unauthorized" is issued after the un-authenticated INVITE and "SIP/2.0 403 Forbidden" after the authenticated INVITE. When I add "insecure=very", this is what the SIP debug shows: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 Found RTP audio format 0 Peer audio RTP is at port 172.16.10.65:36272 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 172.16.10.65:36272 Looking for +15552127020 in from-sip-external (domain sip.acme.com) list_route: hop: It isn't very clear (to me) from the success how the "insecure=very" helps. Frank -Original Message- From: Andres [mailto:and...@telesip.net] Sent: Monday, January 05, 2009 7:43 PM To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very" Frank Bulk - iName.com wrote: >The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not >work unless I add "insecure=very" to my "Outgoing settings", but I don't >want to do that. I do want to authenticate. Outgoing (Asterisk PBX to >Class 5 switch) calls do authenticate and work. > >The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username >and password that it's sending out. But the INVITE is responded by the >Asterisk with "SIP/2.0 403 Forbidden" > >I've changed the INVITE message to mask the real telephone numbers, SIP >server, passwords, and IP addresses, but I did that using search and replace >so the structure is intact. > >What do I need to configure in the "Incoming Settings" panel for the CS >1500's INVITE to my Asterisk server to work? I've tried all kinds of >combinations of user,username,authname using +15552027020,host with IP >and/or DNS name, but nothing appears to work. > > > Do a sip debug on the asterisk console and see if it is actually is matching one of your sip.conf entries during an invite from the CS1500. Look for a line that says something like 'Found Peerbla bla bla'. If you dont see that line, then you are not even adding the correct sip.conf entry to match the invite from the CS1500. Andres http://www.telesip.net >Frank > >INVITE message from Wireshark packet capture: > >INVITE sip:+15552027...@sip.acme.com SIP/2.0 >From: >;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d b >ba4 >To: >Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 >CSeq: 5102 INVITE >Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 >User-Agent: Nortel CS1500UA/v02.00.REL01 >Accept: application/sdp >P-Asserted-Identity: >Privacy: none >Remote-Party-ID: ; party=calling; >privacy=off >Max-Forwards: 70 >Supported: 100rel,replaces >Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK >Contact: >Authorization: Digest >username="username",realm="asterisk",nonce="118af2b0",uri="sip:+15552027020 @ >sip.acme.com",response="111e63ec2a1f3ebabefe4f7dae4087a1",algorithm=MD5 >Content-Type: application/SDP >Content-Length: 167 > >v=0 >o=- 2973921782 2973921782 IN IP4 172.16.10.65 >s=SIP Call >c=IN IP4 172.16.10.65 >t=0 0 >m=audio 36224 RTP/AVP 0 >a=rtpmap:0 PCMU/8000 >a=ptime:20 >a=sendrecv > > >___ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"
Well, it's not "acme.com", but another domain. That information about the encoding process of the nonce is helpful to know. Do I need to specify the context to be "sip.acme.com"? Where is that "acme.com" specified in the trunk configuration? Frank -Original Message- From: Alex Balashov [mailto:abalas...@evaristesys.com] Sent: Monday, January 05, 2009 7:04 PM To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very" Is sip.acme.com actually the domain you want to use? Keep in mind the domain is part of the digest authentication process and is a factor in the encoding of the nonce. Frank Bulk - iName.com wrote: > The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not > work unless I add "insecure=very" to my "Outgoing settings", but I don't > want to do that. I do want to authenticate. Outgoing (Asterisk PBX to > Class 5 switch) calls do authenticate and work. > > The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username > and password that it's sending out. But the INVITE is responded by the > Asterisk with "SIP/2.0 403 Forbidden" > > I've changed the INVITE message to mask the real telephone numbers, SIP > server, passwords, and IP addresses, but I did that using search and replace > so the structure is intact. > > What do I need to configure in the "Incoming Settings" panel for the CS > 1500's INVITE to my Asterisk server to work? I've tried all kinds of > combinations of user,username,authname using +15552027020,host with IP > and/or DNS name, but nothing appears to work. > > Frank > > INVITE message from Wireshark packet capture: > > INVITE sip:+15552027...@sip.acme.com SIP/2.0 > From: > ;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2db > ba4 > To: > Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 > CSeq: 5102 INVITE > Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 > User-Agent: Nortel CS1500UA/v02.00.REL01 > Accept: application/sdp > P-Asserted-Identity: > Privacy: none > Remote-Party-ID: ; party=calling; > privacy=off > Max-Forwards: 70 > Supported: 100rel,replaces > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK > Contact: > Authorization: Digest > username="username",realm="asterisk",nonce="118af2b0",uri="sip:+15552027020@ > sip.acme.com",response="111e63ec2a1f3ebabefe4f7dae4087a1",algorithm=MD5 > Content-Type: application/SDP > Content-Length: 167 > > v=0 > o=- 2973921782 2973921782 IN IP4 172.16.10.65 > s=SIP Call > c=IN IP4 172.16.10.65 > t=0 0 > m=audio 36224 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > a=sendrecv > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"
Frank Bulk - iName.com wrote: >The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not >work unless I add "insecure=very" to my "Outgoing settings", but I don't >want to do that. I do want to authenticate. Outgoing (Asterisk PBX to >Class 5 switch) calls do authenticate and work. > >The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username >and password that it's sending out. But the INVITE is responded by the >Asterisk with "SIP/2.0 403 Forbidden" > >I've changed the INVITE message to mask the real telephone numbers, SIP >server, passwords, and IP addresses, but I did that using search and replace >so the structure is intact. > >What do I need to configure in the "Incoming Settings" panel for the CS >1500's INVITE to my Asterisk server to work? I've tried all kinds of >combinations of user,username,authname using +15552027020,host with IP >and/or DNS name, but nothing appears to work. > > > Do a sip debug on the asterisk console and see if it is actually is matching one of your sip.conf entries during an invite from the CS1500. Look for a line that says something like 'Found Peerbla bla bla'. If you dont see that line, then you are not even adding the correct sip.conf entry to match the invite from the CS1500. Andres http://www.telesip.net >Frank > >INVITE message from Wireshark packet capture: > >INVITE sip:+15552027...@sip.acme.com SIP/2.0 >From: >;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2db >ba4 >To: >Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 >CSeq: 5102 INVITE >Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 >User-Agent: Nortel CS1500UA/v02.00.REL01 >Accept: application/sdp >P-Asserted-Identity: >Privacy: none >Remote-Party-ID: ; party=calling; >privacy=off >Max-Forwards: 70 >Supported: 100rel,replaces >Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK >Contact: >Authorization: Digest >username="username",realm="asterisk",nonce="118af2b0",uri="sip:+15552027020@ >sip.acme.com",response="111e63ec2a1f3ebabefe4f7dae4087a1",algorithm=MD5 >Content-Type: application/SDP >Content-Length: 167 > >v=0 >o=- 2973921782 2973921782 IN IP4 172.16.10.65 >s=SIP Call >c=IN IP4 172.16.10.65 >t=0 0 >m=audio 36224 RTP/AVP 0 >a=rtpmap:0 PCMU/8000 >a=ptime:20 >a=sendrecv > > >___ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"
Is sip.acme.com actually the domain you want to use? Keep in mind the domain is part of the digest authentication process and is a factor in the encoding of the nonce. Frank Bulk - iName.com wrote: > The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not > work unless I add "insecure=very" to my "Outgoing settings", but I don't > want to do that. I do want to authenticate. Outgoing (Asterisk PBX to > Class 5 switch) calls do authenticate and work. > > The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username > and password that it's sending out. But the INVITE is responded by the > Asterisk with "SIP/2.0 403 Forbidden" > > I've changed the INVITE message to mask the real telephone numbers, SIP > server, passwords, and IP addresses, but I did that using search and replace > so the structure is intact. > > What do I need to configure in the "Incoming Settings" panel for the CS > 1500's INVITE to my Asterisk server to work? I've tried all kinds of > combinations of user,username,authname using +15552027020,host with IP > and/or DNS name, but nothing appears to work. > > Frank > > INVITE message from Wireshark packet capture: > > INVITE sip:+15552027...@sip.acme.com SIP/2.0 > From: > ;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2db > ba4 > To: > Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 > CSeq: 5102 INVITE > Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 > User-Agent: Nortel CS1500UA/v02.00.REL01 > Accept: application/sdp > P-Asserted-Identity: > Privacy: none > Remote-Party-ID: ; party=calling; > privacy=off > Max-Forwards: 70 > Supported: 100rel,replaces > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK > Contact: > Authorization: Digest > username="username",realm="asterisk",nonce="118af2b0",uri="sip:+15552027020@ > sip.acme.com",response="111e63ec2a1f3ebabefe4f7dae4087a1",algorithm=MD5 > Content-Type: application/SDP > Content-Length: 167 > > v=0 > o=- 2973921782 2973921782 IN IP4 172.16.10.65 > s=SIP Call > c=IN IP4 172.16.10.65 > t=0 0 > m=audio 36224 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > a=sendrecv > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"
The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add "insecure=very" to my "Outgoing settings", but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with "SIP/2.0 403 Forbidden" I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the "Incoming Settings" panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com SIP/2.0 From: ;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2db ba4 To: Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: Privacy: none Remote-Party-ID: ; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: Authorization: Digest username="username",realm="asterisk",nonce="118af2b0",uri="sip:+15552027020@ sip.acme.com",response="111e63ec2a1f3ebabefe4f7dae4087a1",algorithm=MD5 Content-Type: application/SDP Content-Length: 167 v=0 o=- 2973921782 2973921782 IN IP4 172.16.10.65 s=SIP Call c=IN IP4 172.16.10.65 t=0 0 m=audio 36224 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
