Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
This might help to answer poster's question. It tells how the allow anonymous sip connections work in FreePBX, and shows the code. http://www.geekzone.co.nz/sbiddle/7183 http://www.geekzone.co.nz/sbiddle/7183-- Zeeshan On Sun, Sep 12, 2010 at 12:11 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Sep 11, 2010 at 9:40 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Poster is having problem when he disallows anonymous sip peers. Do you know at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet seen the dialplan for FreePBX. It's very simple to find the actually issue, if the OP does the following: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt The attached the debug log to thread. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 -- Zeeshan A Zakaria -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Thanks guys. I wasn't able to collect enough SIP debug as the problem was resolved as I was testing different configuration for the trunk. Probably a change on the provider side. John Novack: Unfortunately, it seems that this list has a non-stop list of people who like to stir up things or try to censor people who bring legit questions without the consideration that they are not moderators of the list at any level. They forget to remember that AsteriskNow uses FreePBX as well and that Asterisk IS the underlying technology for all the flavours. Thanks for the feedback. Most I was able to collect was that: - if the trunk configuration even included context=from-pstn, the CLI would show Executing .@from-sip-external. - if SIP Anonymous was set to YES then the [from-sip-external] context would match the peer to the right trunk defined as that is what is expected of that context from the code. If SIP Anonymous was set to off @from-sip-external is set to go to ss-noservice. - Later on when the calls resumed and the problem was fixed, calls were coming in with Executing @from-pstn which should have always been the case regardless of the SIP Anonymous or not. I was puzzled because the FISRT line of the CLI was the Executing @from-pstn or Executing .@from-sip-external and that made a world of difference. The latter one not working. I just couldn't pinpoint where FreePBX failed to read the context=from-pstn. If it was something to do with the MySQL database or of parsing the _custom.conf files as the problem was fixed all a sudden. I guess now I have to wait and see if it comes back. Thanks, On Tue, Sep 14, 2010 at 10:47 AM, Zeeshan Zakaria zisha...@gmail.comwrote: This might help to answer poster's question. It tells how the allow anonymous sip connections work in FreePBX, and shows the code. http://www.geekzone.co.nz/sbiddle/7183 http://www.geekzone.co.nz/sbiddle/7183-- Zeeshan On Sun, Sep 12, 2010 at 12:11 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Sep 11, 2010 at 9:40 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Poster is having problem when he disallows anonymous sip peers. Do you know at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet seen the dialplan for FreePBX. It's very simple to find the actually issue, if the OP does the following: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt The attached the debug log to thread. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 -- Zeeshan A Zakaria -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
This is not elastix or FreePBX forum and asking non-asterisk related questions here is misusing this mailing list. Allow anonymous sip is not an asterisk feature. Look in the code in extensions.conf what it is programmed to do and you'll figure out why it is happening. Or maybe post the code and ask why such a behaviour, which'll be better way to ask this elastix related question here. If you know what this part of dialplan does, rest is easy to figure out. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-10 11:17 PM, bruce bruce bruceb...@gmail.com wrote: Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing sip set debug peer PROVIDER: Sending to 123.123.123.123 : 5060 (no NAT) That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see the ACK, REQUEST and ACCEPT of sip debug just fine. This is Elastix with Asterisk 1.4.33.1 Any thoughts? Thanks -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Allow anonymous SIP and enable debug then check if calls coming from same IP which you have configured in peer? Regards, Faisal Hanif// On 9/11/2010 8:07 AM, bruce bruce wrote: Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing sip set debug peer PROVIDER: Sending to 123.123.123.123 : 5060 (no NAT) That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see the ACK, REQUEST and ACCEPT of sip debug just fine. This is Elastix with Asterisk 1.4.33.1 Any thoughts? Thanks -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Zeeshan Zakaria wrote: This is not elastix or FreePBX forum and asking non-asterisk related questions here is misusing this mailing list. Since the poster may not be sure this isn't an Asterisk problem, and Asterisk IS involved, your position is unreasonable. Self appointed list police aren't received well, and waste archive space. If one thinks the post isn't appropriate for the list, simply delete and move on. JN Allow anonymous sip is not an asterisk feature. Look in the code in extensions.conf what it is programmed to do and you'll figure out why it is happening. Or maybe post the code and ask why such a behaviour, which'll be better way to ask this elastix related question here. If you know what this part of dialplan does, rest is easy to figure out. Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com On 2010-09-10 11:17 PM, bruce bruce bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing sip set debug peer PROVIDER: Sending to 123.123.123.123 : 5060 (no NAT) That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see the ACK, REQUEST and ACCEPT of sip debug just fine. This is Elastix with Asterisk 1.4.33.1 Any thoughts? Thanks -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
On Fri, Sep 10, 2010 at 11:07 PM, bruce bruce bruceb...@gmail.com wrote: I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Have you considered contacting your provider? I would think that is your first step. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Mr. John, This is not about policing and this is asterisk-user mailing list. Poster is a FreePBX user. I am very well aware of Asterisk IS involved, but the fact is this is not a FreePBX mailing list. If the poster examines the problem code from extensions.conf, or post it here, it'll made him and everyone clear why is it happening. But poster apparently not well verse in Asterisk anyways. FreePBX has their own forum as well. Or maybe you can explore FreePBX code for him. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-11 9:40 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Sep 10, 2010 at 11:07 PM, bruce bruce bruceb...@gmail.com wrote: I have a provider whose... Have you considered contacting your provider? I would think that is your first step. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Pr... -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
On Fri, 2010-09-10 at 23:07 -0400, bruce bruce wrote: Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing sip set debug peer PROVIDER: Sending to 123.123.123.123 : 5060 (no NAT) That is ALL I am getting with sip debug turned on. I think this may be because the peer is not be recognized as a peer. If you know the IP of the source of the call (the provider) try sip set debug ip X.X.X.X. Then you will probably see the rejection. Not that that will help you much :) You need to find out why it is being rejected. Either you changed the peer parameters or they did... j -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
On Sat, 2010-09-11 at 03:53 -0400, Zeeshan Zakaria wrote: This is not elastix or FreePBX forum and asking non-asterisk related questions here is misusing this mailing list. Allow anonymous sip is not an asterisk feature. Look in the code in extensions.conf what it is programmed to do and you'll figure out why it is happening. Or maybe post the code and ask why such a behaviour, which'll be better way to ask this elastix related question here. If you know what this part of dialplan does, rest is easy to figure out. Zeeshan A Zakaria Heh - listen to you - top posting, bad english, and self appointed list police. His problem certainly seemed asterisk related to me, and has NOTHING to do with code in extensions.conf. He even posted CLI commands he is attempting to use to find his problem. I applaud him for taking the initiative to try working it out on his own, and see no problem at all with his question. I hope we can help him fix it. j -- www.ilovetovoip.com On 2010-09-10 11:17 PM, bruce bruce bruceb...@gmail.com wrote: Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing sip set debug peer PROVIDER: Sending to 123.123.123.123 : 5060 (no NAT) That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see the ACK, REQUEST and ACCEPT of sip debug just fine. This is Elastix with Asterisk 1.4.33.1 Any thoughts? Thanks -- _ -- 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 -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
I think this may be because ... So you think, don't know. Maybe you knew if you knew the FreePBX code, or bothered to look into it. j -- _ -- 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 -- Zeeshan A Zakaria -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
So you are sure it has NOTHING to do with extensions.conf. This clearly
shows your absolute ignorance about what poster is asking and how FreePBX
works. Had the problem code been posted, this problem would already have
been solved by now.
And sorry if you think this is policing. You can think whatever you like.
--
Zeeshan
On Sat, Sep 11, 2010 at 2:43 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Sat, 2010-09-11 at 03:53 -0400, Zeeshan Zakaria wrote:
This is not elastix or FreePBX forum and asking non-asterisk related
questions here is misusing this mailing list. Allow anonymous sip is
not an asterisk feature. Look in the code in extensions.conf what it
is programmed to do and you'll figure out why it is happening. Or
maybe post the code and ask why such a behaviour, which'll be better
way to ask this elastix related question here. If you know what this
part of dialplan does, rest is easy to figure out.
Zeeshan A Zakaria
Heh - listen to you - top posting, bad english, and self appointed list
police. His problem certainly seemed asterisk related to me, and has
NOTHING to do with code in extensions.conf. He even posted CLI commands
he is attempting to use to find his problem. I applaud him for taking
the initiative to try working it out on his own, and see no problem at
all with his question. I hope we can help him fix it.
j
--
www.ilovetovoip.com
On 2010-09-10 11:17 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Everyone,
I have a provider whose DID used to come into the box just fine but
recently stopped working. Nothing has been changed on our end.
Here is what I get when doing sip set debug peer PROVIDER:
Sending to 123.123.123.123 : 5060 (no NAT)
That is ALL I am getting with sip debug turned on.
With Allow Anonymous SIP set to YES, then the call comes in properly
and you see the ACK, REQUEST and ACCEPT of sip debug just fine.
This is Elastix with Asterisk 1.4.33.1
Any thoughts?
Thanks
--
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Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere j...@sunfone.com wrote: Sending to 123.123.123.123 : 5060 (no NAT) Either you changed the peer parameters or they did... If he is not receiving any response, it is most likely a routing issue. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Actually it is a very easy to understand and fix issue, but looking at the code taking care of anonymous sip calls is the key. Those who post third party GUI related issues should at least post the underlying asterisk config or code here, so the asterisk part of the problem can be fixed. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-11 7:22 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere j...@sunfone.com wrote: Sending to 123.123.12... Either you changed the peer parameters or they did... If he is not receiving any response, it is most likely a routing issue. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger ... -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a... -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
-- www.ilovetovoip.com On 2010-09-11 7:22 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere j...@sunfone.com wrote: Sending to 123.123.12... Either you changed the peer parameters or they did... If he is not receiving any response, it is most likely a routing issue. -- [un top posting] On Sat, 2010-09-11 at 19:30 -0400, Zeeshan Zakaria wrote: Actually it is a very easy to understand and fix issue, but looking at the code taking care of anonymous sip calls is the key. Those who post third party GUI related issues should at least post the underlying asterisk config or code here, so the asterisk part of the problem can be fixed. Zeeshan A Zakaria Its not that he isn't receiving a response - its that his peer debug statement isn't getting tripped because the peer hasn't authenticated. That's why I suggested he debug by IP rather than peer. Then what he will see is the SIP auth attempts and asterisk rejecting them, but in my experience not much is of value in seeing those packets - it doesn't point to *why* the connection is being rejected. The routing must be ok since allowing guest sip connections (the result of setting accept anonymous in FreePBX) allows the calls to come in fine. His problem is the peer authenticating. This of course has nothing to do with extensions.conf, as the dialplan is not involved. It is a SIP authentication problem, purely. There is no relevant code to post, and if you had ever looked into FreePBX's relevant code you would realize that it is actually fairly complex, and you would indeed have a difficult time debugging the flow. It *might* help if he posted his peer entry, but without seeing the other side that may not help much either. As Paul suggested first off, he should be in touch with his provider, whose tech support should be able to help him sort it out. I ran into a strange one EXACTLY like this just last week. We have a residential dial-tone customer with a Linksys SPA2102 (our standard device for this service). He had someone come out and replace his home router, and when he did he stopped authenticating. He has a fixed IP, so I enabled the debugging as I have mentioned twice now (by IP) and saw the attempts and rejections. After much hair pulling I *disabled* nat in his peer entry and it suddenly connected fine. This is bizarre, as our standard peer configuration works for 100% of the rest of our customers, who all connect from behind their home nat gateways of all kinds. I still don't know why that fixed it. Sorry you took it so harshly Zeeshan, but the only posts that stick out to me from you are the ones where you are bashing people for posting questions. I don't recall any off the top of my head where you are actually helping. Yup, I consider that policing, and it isn't needed. Like someone else suggested, if you don't want to read it, delete it. And no, I am not going to bother to read back through archives to see if that is the truth. Its my impression of your posts, thats all. j -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Poster is having problem when he disallows anonymous sip peers. Do you know at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet seen the dialplan for FreePBX. When there is enough detail in the post and I am aware of the problem, I always try to help. I don't believe in making guesses. Troubleshooting requires some good detail of the problem. And yes, answering non-asterisk related issues is not the goal of this mailing list. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-11 9:24 PM, Jeff LaCoursiere j...@sunfone.com wrote: -- www.ilovetovoip.com On 2010-09-11 7:22 PM, Paul Belanger paul.belan...@polybea... [un top posting] On Sat, 2010-09-11 at 19:30 -0400, Zeeshan Zakaria wrote: Actually it is a very easy to understan... Its not that he isn't receiving a response - its that his peer debug statement isn't getting tripped because the peer hasn't authenticated. That's why I suggested he debug by IP rather than peer. Then what he will see is the SIP auth attempts and asterisk rejecting them, but in my experience not much is of value in seeing those packets - it doesn't point to *why* the connection is being rejected. The routing must be ok since allowing guest sip connections (the result of setting accept anonymous in FreePBX) allows the calls to come in fine. His problem is the peer authenticating. This of course has nothing to do with extensions.conf, as the dialplan is not involved. It is a SIP authentication problem, purely. There is no relevant code to post, and if you had ever looked into FreePBX's relevant code you would realize that it is actually fairly complex, and you would indeed have a difficult time debugging the flow. It *might* help if he posted his peer entry, but without seeing the other side that may not help much either. As Paul suggested first off, he should be in touch with his provider, whose tech support should be able to help him sort it out. I ran into a strange one EXACTLY like this just last week. We have a residential dial-tone customer with a Linksys SPA2102 (our standard device for this service). He had someone come out and replace his home router, and when he did he stopped authenticating. He has a fixed IP, so I enabled the debugging as I have mentioned twice now (by IP) and saw the attempts and rejections. After much hair pulling I *disabled* nat in his peer entry and it suddenly connected fine. This is bizarre, as our standard peer configuration works for 100% of the rest of our customers, who all connect from behind their home nat gateways of all kinds. I still don't know why that fixed it. Sorry you took it so harshly Zeeshan, but the only posts that stick out to me from you are the ones where you are bashing people for posting questions. I don't recall any off the top of my head where you are actually helping. Yup, I consider that policing, and it isn't needed. Like someone else suggested, if you don't want to read it, delete it. And no, I am not going to bother to read back through archives to see if that is the truth. Its my impression of your posts, thats all. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.aste... -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
[snip] customers, who all connect from behind their home nat gateways of all kinds. I still don't know why that fixed it. Sorry you took it so harshly Zeeshan, but the only posts that stick out to me from you are the ones where you are bashing people for posting questions. I don't recall any off the top of my head where you are actually helping. Yup, I consider that policing, and it isn't needed. Like someone else suggested, if you don't want to read it, delete it. And no, I am not going to bother to read back through archives to see if that is the truth. Its my impression of your posts, thats all. j [un top posting again] On Sat, 11 Sep 2010, Zeeshan Zakaria wrote: Poster is having problem when he disallows anonymous sip peers. Do you know at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet seen the dialplan for FreePBX. Umm, no, poster is having problems that are only SOLVED by allowing guest sip connections (since you want to stick with asterisk terms, not FreePBX, right?). That is because when he doesn't allow guest connections his inbound calls are getting rejected, as they are not matching any of his defined peers. I'm not guessing here - these are facts based on his observations. Your bizarre assumptions that he (or I) need to better understand FreePBX's dialplan code are guesses. j-- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
On Sep 10, 2010, at 10:07 PM, bruce bruce bruceb...@gmail.com wrote: Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing sip set debug peer PROVIDER: Sending to 123.123.123.123 : 5060 (no NAT) That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see the ACK, REQUEST and ACCEPT of sip debug just fine. This is Elastix with Asterisk 1.4.33.1 Any thoughts? Thanks Try adding 'insecure=port,invite' to the sip peer definition. If that doesn't work, you can try 'insecure=very'. Otherwise, try getting a packet capture using tcpdump on the interface that you world normally connect to your provider with in order to see all of the sip traffic that you're missing with just a peer debug from within asterisk. Once you have that, if the answer doesn't jump out at you, you can post it here and someone should be able to help. Thanks, --Warren Selby -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
On 09/12/10 07:06, Zeeshan Zakaria wrote: I think this may be because ... So you think, don't know. Maybe you knew if you knew the FreePBX code, or bothered to look into it. For God's sake, stick a sock in it. Others are attempting to help. You are not. -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
On Sat, Sep 11, 2010 at 9:40 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Poster is having problem when he disallows anonymous sip peers. Do you know at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet seen the dialplan for FreePBX. It's very simple to find the actually issue, if the OP does the following: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt The attached the debug log to thread. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing sip set debug peer PROVIDER: Sending to 123.123.123.123 : 5060 (no NAT) That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see the ACK, REQUEST and ACCEPT of sip debug just fine. This is Elastix with Asterisk 1.4.33.1 Any thoughts? Thanks -- _ -- 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
