Re: [asterisk-users] SIP account registration fails after upgrade to 1.8
-Original Message- From: Jaap Winius jwin...@umrk.nl Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP account registration fails after upgrade to 1.8 Date: Fri, 22 Mar 2013 02:46:43 + (UTC) On Thu, 21 Mar 2013 16:35:16 +, Jaap Winius wrote: Hopefully, my ISP will see fit to squash this bug ASAP. Well, I got my answer from them quickly enough: Nope. Luckily, somebody was kind enough to suggest a workaround. Unfortunately, it involves, downloading the source code and making a few changes to it to prevent Asterisk from adding '@IPaddress' to the end of the Call-ID string. Nevertheless, it's easy enough to do. The idea is to look for this string that appears twice in ./channels/chan_sip.c: ast_string_field_build(pvt, callid, %s@%s, generate_random_string(buf, sizeof(buf)), host); And to change it to: ast_string_field_build(pvt, callid, %s, generate_random_string(buf, sizeof(buf))); Now my Call-IDs look like this: Call-ID: 63935a8d2144d4f1309024fd7612f608 Instead of this: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a] Still, I'd much prefer that my ISP fixed the problem instead, because now every time a security update becomes available for Asterisk, I'm going to have to download the source code, make the same changes, recompile it and install it all over again and again. Ho hum. Of course, an even better solution would be if Asterisk had a variable with which to alter the Call-ID string format so that I could omit the IP address. :-) Cheers, Jaap -Original Message- Hi Jaap, just wondering, might this perhaps be an IPv6 quirk? By altering '@IPaddress' you got rid of : '@[2001:888:abcd:1::a]' Does the dame happen with V4-only? I presume you didn't activate V6 at your end lately? Other idea (perhaps pointless), you got the numeric address, would the same issue still exists if '2001:888:abcd:1::a' could be translater back into a dns-name? (include it in your /etc/hosts ?) Sometimes the '[]' cause some side-effects (specially if some regex are used unseen) Groet, hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP account registration fails after upgrade to 1.8
On Thu, 21 Mar 2013 16:35:16 +, Jaap Winius wrote: ... For example, if my server sends it a SIP packet with a register request and a Call-ID that looks like this: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a] ... somewhere along they line they end up changing it to this: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:ABCD:1::A] Actually, I have to correct myself here. Not only was the SIP server at sip.xs4all.nl changing the lower case letters of the IPv6 section in any Call-IDs to upper case, it was also expanding the addresses, like so: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:ABCD:1:0:0:0:A] So, that SIP server was (and still is as of this writing) actually making two mistakes instead of just one. My apologies for not being entirely accurate the first time around. Cheers, Jaap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP account registration fails after upgrade to 1.8
On Fri, 22 Mar 2013 02:46:43 +, Jaap Winius wrote: Of course, an even better solution would be if Asterisk had a variable with which to alter the Call-ID string format so that I could omit the IP address. :-) It turns out that there in a variable that can do exactly that, and is therefore the solution to this problem: 'fromdomain=domain_name'. Once placed in the [general] section of your sip.conf, Asterisk will generate Call-IDs for its SIP packets that end with an '@' followed by your chosen domain name instead of your server's IPv6 address. Thanks to Rob van der Putten for this solution! Cheers, Jaap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP account registration fails after upgrade to 1.8
On Tue, 19 Mar 2013 02:15:10 +, Jaap Winius wrote: Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9 to 1.8.13, my server is no longer able to register a connection to a SIP account at my ISP (XS4ALL in the Netherlands). At the same time, it is still able to register a different account with another SIP provider... To answer my own question, this turned out to be due to a bug in the SIP server at XS4ALL. I discovered it after using tcpdump to examine the exchange of packets during my registration attempts and noticing that Asterisk 1.8.13.1 was using an IPv6 address in the Call-ID instead of an IPv4 address as before. According to the specification for SIP 2.0 (RFC 3261) this is perfectly legal, just as long as both parties treat the entire Call-ID as a string and never make any changes to it. However, I discovered that is was exactly what the SIP server at XS4ALL is doing. For example, if my server sends it a SIP packet with a register request and a Call-ID that looks like this: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a] ... somewhere along they line they end up changing it to this: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:ABCD:1::A] In other words, it is treating the latter part of the Call-ID not as a string, but as an IPv6 address and has taken it upon itself to change all of the letters in that address to upper case. This changes the Call-ID and thus my registration attemp cannot be completed. Of course, this won't affect you if you happen to have an IPv6 address without any letters in it. This situation is in contrast to another SIP provider that I use, sip.internetcalls.com, with which I currently have no problems because they leave such Call-IDs unchanged. I don't know what kind of SIP server software they use, but XS4ALL appears to be using Cirpack 4.42a. This bug is very similar to another one described in this forum exchange: http://forums.asterisk.org/viewtopic.php?f=1t=84603start=0 Here, a SIP server at an ISP was taking the IPv6 address at the end of a Call-ID and expanding it, e.g. from ::1 (the IPv6 loopback address) to 0:0:0:0:0:0:0:1. In both that case and in mine, we get the same result: an altered Call-ID that leads to endless timeouts and no registration. Hopefully, my ISP will see fit to squash this bug ASAP. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP account registration fails after upgrade to 1.8
On Thu, 21 Mar 2013 16:35:16 +, Jaap Winius wrote: Hopefully, my ISP will see fit to squash this bug ASAP. Well, I got my answer from them quickly enough: Nope. Luckily, somebody was kind enough to suggest a workaround. Unfortunately, it involves, downloading the source code and making a few changes to it to prevent Asterisk from adding '@IPaddress' to the end of the Call-ID string. Nevertheless, it's easy enough to do. The idea is to look for this string that appears twice in ./channels/chan_sip.c: ast_string_field_build(pvt, callid, %s@%s, generate_random_string(buf, sizeof(buf)), host); And to change it to: ast_string_field_build(pvt, callid, %s, generate_random_string(buf, sizeof(buf))); Now my Call-IDs look like this: Call-ID: 63935a8d2144d4f1309024fd7612f608 Instead of this: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a] Still, I'd much prefer that my ISP fixed the problem instead, because now every time a security update becomes available for Asterisk, I'm going to have to download the source code, make the same changes, recompile it and install it all over again and again. Ho hum. Of course, an even better solution would be if Asterisk had a variable with which to alter the Call-ID string format so that I could omit the IP address. :-) Cheers, Jaap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP account registration fails after upgrade to 1.8
hi, try srvlookup=yes On Tue, Mar 19, 2013 at 3:15 AM, Jaap Winius jwin...@umrk.nl wrote: Hi folks, Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9 to 1.8.13, my server is no longer able to register a connection to a SIP account at my ISP (XS4ALL in the Netherlands). At the same time, it is still able to register a different account with another SIP provider, so it must be that they no longer have the same basic requirements. The relevant part of my sip.conf looks like this: [general] context=incoming-j canreinvite=no dtmfmode=inband qualify=yes srvlookup=no disallow=all allow=alaw allow=ulaw allow=g722 allow=g726 allow=g729 insecure=port,invite register = telno:password@sip.xs4all.nl/telno Does anyone know of any new variables that have been introduced since Asterisk 1.6.2.9, that apply here and might be causing this problem? Thanks, Jaap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP account registration fails after upgrade to 1.8
On Tue, 19 Mar 2013 11:58:22 +0100, Asghar Mohammad wrote: try srvlookup=yes Already tried that, but enabling DNS lookups makes no difference when establishing the SIP connection. The error message that I keep seeing at the console looks like this: [Mar 19 12:47:21] NOTICE[7494]: chan_sip.c:13171 sip_reg_timeout:-- Registration for 'telno@sip.xs4all.nl' timed out, trying again (Attempt #3) Incidentally, I have remote access to two other Asterisk systems in the Netherlands with XS4ALL connections, both still Debian squeeze with Asterisk 1.6.2.9, and when I add my register line to their sip.conf files, which are virtually identical to mine (except for the context), it registers immediately. This shows that my account still works. Moreover, I also have remote access to some more Asterisk systems with XS4ALL connections and Debian wheezy with Asterisk 1.8.13.1. When I add my register line to their sip.conf files, which are virtually identical (except for the context), it fails. For the rest those systems are pretty much like my own, but at least it demonstrates that the problem is not unique to my system and connection. Oh, and all of these systems have srvlookup=no (default is yes). Thanks anyway, Jaap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP account registration fails after upgrade to 1.8
Hi folks, Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9 to 1.8.13, my server is no longer able to register a connection to a SIP account at my ISP (XS4ALL in the Netherlands). At the same time, it is still able to register a different account with another SIP provider, so it must be that they no longer have the same basic requirements. The relevant part of my sip.conf looks like this: [general] context=incoming-j canreinvite=no dtmfmode=inband qualify=yes srvlookup=no disallow=all allow=alaw allow=ulaw allow=g722 allow=g726 allow=g729 insecure=port,invite register = telno:password@sip.xs4all.nl/telno Does anyone know of any new variables that have been introduced since Asterisk 1.6.2.9, that apply here and might be causing this problem? Thanks, Jaap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
