Re: [asterisk-users] Basic Sip.conf and extensions.conf
enable sip debug and check which error or error code you are getting also try nat=yes On Mon, Jan 17, 2011 at 5:34 PM, Thomas Perron thomas.per...@gmail.comwrote: Thanks. I fixed that. Still does not work. On Mon, Jan 17, 2011 at 12:53 AM, Jeroen Eeuwes jeroeneeu...@gmail.com wrote: Hi Thomas, register = 999:999...@sip.callwithus.comi Perhaps this should be .com instead of .comi ? Best regards, Jeroen Eeuwes -- _ -- 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 -- Regards, Zuhair Raza Asst. Manager Technical Support Phone: +1-850-433-8555 ext 101064 Website: www.didx.net Skype: zuhairraza What is DIDX.net? http://www.youtube.com/watch?v=mIgGTGkTZns http://www.youtube.com/watch?v=mIgGTGkTZns -- _ -- 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] Ongoing problem with 1.8
On Tuesday 18 January 2011 01:05:20 Ira wrote: I have tried installing many of the beta versions and most of the release versions of 1.8. I have 3 SIP phones which we use for all our calls. After installing 1.8 the first thing I try is calling out port one of my Digium TDM04 back into port 2. I can see that the call comes in and tries to call all three SIP phones but the phones never ring. Eventually the call goes to voice mail and these error messages pop up. I've read doc/sip-retransmit.txt and as far as I can tell, there's nothing there for me to try. Is there anything else I might try or do to help troubleshoot this. Try running a tcpdump for udp port 5060 while this is occurring. Also, what type of SIP phones are you using? -- Tilghman -- _ -- 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] Top Posting
Top posting? Who cares? Get a life! Now - can we get back to Asterisk et al? Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Murawski Sent: 18 January 2011 02:57 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Top Posting On 01/17/2011 08:26 PM, Matt Riddell wrote: On 17/01/11 4:29 PM, jon pounder wrote: Surely there is some mail client smart enough to be able to flip around the levels of indenting so most recent is top or bottom. If not quit bitching and make one - I will continue top posting since I don't seem to be alone in preferring it. That was one of the first things that came to mind. I'm definitely more keen on inline replies - if you reply to 20 points in someone's email you quote the part you're replying to then reply to it. That was the standard for much of the 90's for emails. I do like that method but most people don't seem to do it anymore. In a long email it's the only way. Otherwise you'd scroll down to find the question, scroll up to find the answer, scroll down to find the next question, scroll up for the next answer etc - crazy. It's also easier to keep the context of what's going on. If replying in one big block, I try to keep the style of one paragraph of response for each paragraph of question, but sometimes stuff just mixes in between and you can easily lose context. Much easier when replies are inline with the questions. It gets hard to follow when there's a dozen nested levels of reply. In conclusion, I think it just depends (tm). -- _ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Sound quality issue
Something that often gets forgotten is the on-site LAN infrastructure as well. It could be a bad/faulty switch, rubbish cabling, induced interference etc. etc. all at the customers premises. Maybe a handset plugged directly in to the back of the router, before it hits the LAN would tell you whether the call is actually getting 'distorted' en-route or not? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: 16 January 2011 12:28 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sound quality issue Le 15/01/2011 20:38, Cédric Lemarchand a écrit : Hello, Hi [...] I am sure there are RTP packets losses somewhere, except RTP debug in the asterisk CLI, how can i determine where the problem come from ? [...] You don't tell which protocol (SIP, IAX, H323) nor which asterisk version. FYI Asterisk 1.6.2.15 in iax had audio quality problems, solved in 1.6.2.16. If you have the possibility, connect directly a phone to the server, eg Device - LAN (no MPLS) - Asterisk. Check also if a call to echo test has the same bad quality. -- Daniel -- _ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Can I know if a call is transffered to asterisk
Good morning My situation is as folowing I have a numer that connect to my asterisk I configured another phone to transfer to this number So when somebody call me he will be transffered to the number which asterisk connect to i.e my asterisk connected phone is not the originated number My question is: How can I know if the call has been transfered to me and what was the number originated by the caller. Regards -- _ -- 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] Can I know if a call is transffered to asterisk
On 01/18/2011 04:41 AM, ishagh ouldbah wrote: Good morning My situation is as folowing I have a numer that connect to my asterisk I configured another phone to transfer to this number So when somebody call me he will be transffered to the number which asterisk connect to i.e my asterisk connected phone is not the originated number My question is: How can I know if the call has been transfered to me and what was the number originated by the caller. If you are receiving the call over an ISDN PRI, and the PRI provider includes RDNIS information, you can inspect that from the dialplan to get the 'redirecting number' (the number that redirected the call to you). Note that this is *not* a transfer (which is a manually initiated operation), but a call forward/redirection. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] how to read mp3
yes i want to know how can i do in order to read this files using apche 2011/1/17 Steve Edwards asterisk@sedwards.com On Mon, 17 Jan 2011, salaheddine elharit wrote: i have asterisk installed in our call centre and I have all the clients conversation saved in this file /usr/apache-tomcat-5.5.17/webapps/aheevaccs/recordings/nosales i have created i php code in order to read this files from server when i put the file in www folder i can read this mp3 files without any issue but when i try to read this files from usr folder my question how can i do in order to read mp3 files form usr folder Are you asking how to read the files using Apache or Asterisk? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Do I need a sip proxy?
Hello Bruce, Sorry for the delay. I don't really have time to follow this list much. In your original setup, you did use a sort of SIP Proxy (the central Asterisk feeding the others) depending on your definition. A SIP Proxy would probably solve your issue, but as I stated in my previous mail, you should not need one. Fixing (or exchanging) Pfsense should also solve your issue and then you'll have one less device that can bring your system down. Fixing Pfsense will probably require you to troubleshoot the issue some more to see exactly what happens, so you know what you need to fix. Compare the SIP traffic between your Asterisks and Pfsense to the traffic between Pfsense and your provider. Capture the traffic in .pcap format with ngrep, tcpdump, wireshark or other packet dumping tools, then analyze it in wireshark. To capture traffic outside Pfsense, you'll probably need to mirror a switch port, install a hub or ask your provider to send you a dump. This will require some understanding of the SIP message format and TCP/IP, but it should not be very complicated. I'm quite sure Pfsense changes the contents of the SIP message itself in ways it should not do possibly in addition to changing the IP packets in ways it should not do. It may also possibly block incoming traffic it should not block. If you decide to use a SIP proxy, then going back to your original design (using Asterisk as a proxy) would probably be the easiest for you. Of the alternatives you've listed, I only have experience with Kamailio. In simple setups, its default configuration will not need to be altered much to get it working. Its logic is VERY different to Asterisk, though. I know that Kamailio would be a very good choice for this role. I believe the alternatives would be as well. With kind regards, Pan B. Christensen Senior technician Ibidium AS http://www.ibidium.no/ - Original Message - From: Bruce B To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, January 11, 2011 4:37 PM Subject: Re: [asterisk-users] Do I need a sip proxy? Thanks a lot for the great input Pan. I think you are right on point with this one. I have STATIC PORT enabled in my outbound WAN. I am not sure if it was set for SIP or OpenVPN use but it is there for a reason. So, I try to mingle a bit with Siproxd package. I am a bit fuzzy on it though. If I have the Siproxd enabled, does it act as a one single server that connects multiple times to my provider or providers and then I connect to the Siproxd in return? Or, I can still register from Asterisk directly with the provider(s) and Siproxd will take care of the SIP packets to be handled nicely? If it's the latter then it sounds fine to use otherwise it would not only be complicated but also a downtime to Siproxd mean downtime to all Asterisk servers. ***In addition I have setup Siproxd according to pfsense guide online but once I save the configurations and return to it there are no configs left. I know this question is for pfsense forum but maybe someone else experienced this? ***And to return to my original question, do I need a SIP proxy and which one would be suit my needs? I still like to get an input on my previous e-mail. I have to stay with pfsense for now as it has proven to be a good router in all other aspect. Thanks, On Tue, Jan 11, 2011 at 7:38 AM, Pan B. Christensen p...@ibidium.no wrote: Hello Bruce, Your understanding of NAT is correct, and your setup should work. I’m not familiar with Pfsense, but I suspected that your problem was due to a SIP ALG. Pfsense seems to have a SIP ALG and other special handling of VoIP traffic. Hence, you are not using plain NAT. Pfsense is probably rewriting the SIP packets in addition to the IP packets. Try reconfiguring Pfsense or swapping it for something else. A good way to troubleshoot your scenario is to compare the traffic in your end to the traffic on your providers end (or on either side of pfsense). Pay attention to the source and destination IP and ports in addition to the contents of the SIP messages. http://doc.pfsense.org/index.php/VoIP_Configuration http://en.wikipedia.org/wiki/Application-level_gateway With kind regards, Pan From: Bruce B Sent: Tuesday, January 11, 2011 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Do I need a sip proxy? Hi Everyone, I am running multiple instances of Asterisk in Proxmox and so far I had one central Asterisk feeding all others with trunks from one provider. Now, I want to connect each Asterisk server directly to the provider. Based on my understanding, each connection made to the provider port 5060 would be on a port that is unique to that server. And so other connections made to the same provider will go out through a different port and should receive responses through that different
Re: [asterisk-users] res_fax_digium.so crashing
On 01/16/2011 09:18 PM, Jeremy Kister wrote: On 1/16/2011 4:13 PM, Paul Belanger wrote: I don't believe Digium is blind to its users: Users of Free Fax For Asterisk are not entitled to any Digium technical support [1]. I'm not looking for technical support; I'm just looking for a way to report a bug and possibly help debug/resolve it. But as you know, Digium's website gives FFA users no clear to contact them - even to report problems. issues.asterisk.org has no selection for res_fax_digium since it is not bundled with Asterisk. I call that willful blindness. Don't get me wrong, I'm grateful for FFA and Asterisk in general - I have several running 1.8.2 working correctly. Alternatively, you can generating an unoptimized backtrace [2] and posting the results to the mailing list, seeing if any member of the community has also had an issue. I didnt expect anyone on this list to be interested, but I suppose you're right. This weekend, i set up a new system running Asterisk 1.8.2 on Debian 5.0.7 where the benchmark told me to use res_fax_digium-1.8.0_1.2.1-core2_32 (i also tried generic_32 but that crashed as well). Asterisk correctly detects the fax and transfers to the fax context. But moments after ReceiveFax is called, asterisk crashes, with no tif written where I've directed it to. I have several files including backtraces and config files at http://jeremy.kister.net/tmp/fax/ We have determined the cause of this problem; I'm sorry to say it's an incompatibility between releases that wasn't caught before the releases were made. res_fax.c in Asterisk 1.8.2 contains some changes that make it incompatible with res_fax_digium modules compiled for Asterisk 1.8.0 and 1.8.1, which is what is causing this problem. For now, the res_fax_digium 1.2.x modules are only usable with Asterisk 1.8.0 and 1.8.1; there is a res_fax_digium 1.3.0 release currently being tested for release that is compatible with Asterisk 1.8.2 and later. We'll try to get it out as quickly as we can, and also update the download selector on the Digium website to indicate that the 1.2.x modules should not be used with Asterisk 1.8.2. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] how to read mp3
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Tuesday, January 18, 2011 6:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] how to read mp3 yes i want to know how can i do in order to read this files using apche 2011/1/17 Steve Edwards asterisk@sedwards.com On Mon, 17 Jan 2011, salaheddine elharit wrote: i have asterisk installed in our call centre and I have all the clients conversation saved in this file /usr/apache-tomcat-5.5.17/webapps/aheevaccs/recordings/nosales i have created i php code in order to read this files from server when i put the file in www folder i can read this mp3 files without any issue but when i try to read this files from usr folder my question how can i do in order to read mp3 files form usr folder Are you asking how to read the files using Apache or Asterisk? In that case, you are in the wrong place, but you just need to associate mp3 with an app on your computer. -- _ -- 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] Asterisk stops responding
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Saturday, January 15, 2011 2:02 AM To: Asterisk Subject: [asterisk-users] Asterisk stops responding I am having a problem with an Asterisk 1.6.2.15 server that runs a small call center with Queuemetrics. In the past month we've had this problem 3 times. The problem is that Asterisk simply stops responding. No calls in or out and you cannot even get to the CLI. The process seems to be running but there is simple no activity. All I see in the log files is: It might be this - I had something similar in behavior though I don't know if I ever got the same error message: https://issues.asterisk.org/view.php?id=18031 -- _ -- 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] Top Posting
On 11-01-18 04:22 AM, Andrew Thomas wrote: Top posting? Who cares? Get a life! Clearly not you, so why both even replying? At worst case it is just redundant information for people, best case somebody reads the email thread at starts bottom posting. I suggest taking a moment and re-reading the thread. If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. Additionally, do you really need a 17 line[1] signature? [1] - http://s3.amazonaws.com/theoatmeal-img/comics/email/4.png -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Do I need a sip proxy?
Thanks for the info. I did get it working without any SIP Proxy. There is a bug in pfSense v1.2.3 where certain configs are not removed and some inconsistencies exist in the xml config file. Once I cleaned that and when I limited my Asterisk servers to use different port ranges for UDP traffic now everything is working great. On Tue, Jan 18, 2011 at 7:26 AM, Pan B. Christensen p...@ibidium.no wrote: Hello Bruce, Sorry for the delay. I don't really have time to follow this list much. In your original setup, you did use a sort of SIP Proxy (the central Asterisk feeding the others) depending on your definition. A SIP Proxy would probably solve your issue, but as I stated in my previous mail, you should not need one. Fixing (or exchanging) Pfsense should also solve your issue and then you'll have one less device that can bring your system down. Fixing Pfsense will probably require you to troubleshoot the issue some more to see exactly what happens, so you know what you need to fix. Compare the SIP traffic between your Asterisks and Pfsense to the traffic between Pfsense and your provider. Capture the traffic in .pcap format with ngrep, tcpdump, wireshark or other packet dumping tools, then analyze it in wireshark. To capture traffic outside Pfsense, you'll probably need to mirror a switch port, install a hub or ask your provider to send you a dump. This will require some understanding of the SIP message format and TCP/IP, but it should not be very complicated. I'm quite sure Pfsense changes the contents of the SIP message itself in ways it should not do possibly in addition to changing the IP packets in ways it should not do. It may also possibly block incoming traffic it should not block. If you decide to use a SIP proxy, then going back to your original design (using Asterisk as a proxy) would probably be the easiest for you. Of the alternatives you've listed, I only have experience with Kamailio. In simple setups, its default configuration will not need to be altered much to get it working. Its logic is VERY different to Asterisk, though. I know that Kamailio would be a very good choice for this role. I believe the alternatives would be as well. With kind regards, Pan B. Christensen Senior technician Ibidium AS http://www.ibidium.no/ - Original Message - *From:* Bruce B bruceb...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Tuesday, January 11, 2011 4:37 PM *Subject:* Re: [asterisk-users] Do I need a sip proxy? Thanks a lot for the great input Pan. I think you are right on point with this one. I have STATIC PORT enabled in my outbound WAN. I am not sure if it was set for SIP or OpenVPN use but it is there for a reason. So, I try to mingle a bit with Siproxd package. I am a bit fuzzy on it though. If I have the Siproxd enabled, does it act as a one single server that connects multiple times to my provider or providers and then I connect to the Siproxd in return? Or, I can still register from Asterisk directly with the provider(s) and Siproxd will take care of the SIP packets to be handled nicely? If it's the latter then it sounds fine to use otherwise it would not only be complicated but also a downtime to Siproxd mean downtime to all Asterisk servers. ***In addition I have setup Siproxd according to pfsense guide online but once I save the configurations and return to it there are no configs left. I know this question is for pfsense forum but maybe someone else experienced this? ***And to return to my original question, do I need a SIP proxy and which one would be suit my needs? I still like to get an input on my previous e-mail. I have to stay with pfsense for now as it has proven to be a good router in all other aspect. Thanks, On Tue, Jan 11, 2011 at 7:38 AM, Pan B. Christensen p...@ibidium.nowrote: Hello Bruce, Your understanding of NAT is correct, and your setup should work. I’m not familiar with Pfsense, but I suspected that your problem was due to a SIP ALG. Pfsense seems to have a SIP ALG and other special handling of VoIP traffic. Hence, you are not using plain NAT. Pfsense is probably rewriting the SIP packets in addition to the IP packets. Try reconfiguring Pfsense or swapping it for something else. A good way to troubleshoot your scenario is to compare the traffic in your end to the traffic on your providers end (or on either side of pfsense). Pay attention to the source and destination IP and ports in addition to the contents of the SIP messages. http://doc.pfsense.org/index.php/VoIP_Configuration http://en.wikipedia.org/wiki/Application-level_gateway With kind regards, Pan *From:* Bruce B bruceb...@gmail.com *Sent:* Tuesday, January 11, 2011 8:58 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Subject:* [asterisk-users] Do I need a sip proxy? Hi Everyone, I
Re: [asterisk-users] Top Posting
Why do I top post? Simple. I read every message in the thread - and if there are 10 messages (for example) in that thread - then why should I have to read them all over again on the last one? Top posting is here - to stay! Stop being so anal and 'retro'. Bottom posting belongs in forums - top post belongs in e-mail lists. There - said it! As for my sig/disclaimer - how about 10 copies of it before you get a reply? That's what bottom posting would have done for you! Anyway Digium, Inc. | Software Developer means you should be developing software - not replying to inane posts like mine :P Have a nice day! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 18 January 2011 14:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting On 11-01-18 04:22 AM, Andrew Thomas wrote: Top posting? Who cares? Get a life! Clearly not you, so why both even replying? At worst case it is just redundant information for people, best case somebody reads the email thread at starts bottom posting. I suggest taking a moment and re-reading the thread. If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. Additionally, do you really need a 17 line[1] signature? [1] - http://s3.amazonaws.com/theoatmeal-img/comics/email/4.png -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Top Posting
I'm top posting this so you will see it and if you don't understand it, look it up. PLONK!! On Tue, 18 Jan 2011, Andrew Thomas wrote: Why do I top post? Simple. I read every message in the thread - and if there are 10 messages (for example) in that thread - then why should I have to read them all over again on the last one? Top posting is here - to stay! Stop being so anal and 'retro'. Bottom posting belongs in forums - top post belongs in e-mail lists. There - said it! As for my sig/disclaimer - how about 10 copies of it before you get a reply? That's what bottom posting would have done for you! Anyway Digium, Inc. | Software Developer means you should be developing software - not replying to inane posts like mine :P Have a nice day! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 18 January 2011 14:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting On 11-01-18 04:22 AM, Andrew Thomas wrote: Top posting? Who cares? Get a life! Clearly not you, so why both even replying? At worst case it is just redundant information for people, best case somebody reads the email thread at starts bottom posting. I suggest taking a moment and re-reading the thread. If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. Additionally, do you really need a 17 line[1] signature? [1] - http://s3.amazonaws.com/theoatmeal-img/comics/email/4.png -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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 Vince. -- Michigan VHF Corp. http://www.nobucks.net/ http://www.CDupe.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] Top Posting
On Tue, 2011-01-18 at 15:18 +, Andrew Thomas wrote: Why do I top post? Simple. I read every message in the thread - and if there are 10 messages (for example) in that thread - then why should I have to read them all over again on the last one? Top posting is here - to stay! Stop being so anal and 'retro'. Bottom posting belongs in forums - top post belongs in e-mail lists. There - said it! As for my sig/disclaimer - how about 10 copies of it before you get a reply? That's what bottom posting would have done for you! Anyway Digium, Inc. | Software Developer means you should be developing software - not replying to inane posts like mine :P Have a nice day! You really took the douche comic to heart. -- With best regards, Fred http://qxork.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] Asterisk SlackBuilds for Slackware Linux
Hello List, To whom it might concern: I have been working in some SlackBuilds (script for making Slackware Packages) for my personal use, but thought they might be useful for someone else here. Beside of the exceptional distributions used so far (CentOS, Debian, Ubuntu, etc.), you might want to test Asterisk on a Slackware Linux box, as it offers outstanding stability and flexibility as well. I. The scripts are located at: http://packages.eslackware.com/slackbuilds/asterisk/ II. So far, they consist of the following: (Listed in recommended building order) a. newt-SlackBuild.tar.gz NewT Libraries b. libpri-SlackBuild.tar.gz LibPRI c. libss7-SlackBuild.tar.gz LibSS7 d. dahdi_linux-SlackBuild.tar.gz DAHDI Linux e. dahdi_tools-SlackBuild.tar.gz DAHDI Tools f. unixODBC-SlackBuild.tar.gz UnixODBC (entire ODBC API, Drivers, and tools) g. mysql_connector_odbc-SlackBuild.tar.gz Connector/ODBC (standardized database driver) h. ptlib-SlackBuild.tar.gz PTLib (used be called PWLib), latest version i. h323plus-SlackBuild.tar.gz H323 Plus (formerly known as OpenH323) j. asterisk-SlackBuild.tar.gz Asterisk 1.4.XX k. asterisk_addons-SlackBuild.tar.gz Asterisk Add-Ons l. template-SlackBuild.tar.gz The SlackBuild template I use for this purposes. You might want to use it for something else. III. Aditional notes: About the scripts: -- You can (and as a good security practice, always should) inspect the scripts, and (if you want so) modify/add/remove parts to them in order to fit your needs. Everything has been tested in Slackware 13.0 and Slackware 13.1 About ODBC + MySQL Connector and Asterisk: -- I needed this feature, but in case you don't, just don't compile/install them and modify the Asterisk SlackBuild in order not enable it. (Personally, I consider the Voicemail ODBC and func_odbc features awesome, give them a try if you can ;) ) About H323 Plus and PTLib: -- The SlackBuilds are written for the latest versions of both, even if Asterisk (at least 1.4.XX) ask for older versions. They compile and install just fine. Problems related to compiling Asterisk using these latest versions were solved as well. About Asterisk: --- If you check the scripts, you will see that everything is as clean as ./configure, make, make install, with some little exceptions in this case: 1. The SlackBuild subtle modifies the configure script in order to make it use your installed (no matter what it is) version of PTLib. 2. The chan_h323 driver is compiled by default (modify the script if you don't want/need it) 3. ODBC support is compiled by default. (again, if you dont wan't/need it, modify the script) Special thanks to Paul Belanger, Tzafrir Cohen and the people at #asterisk in Freenode. PS. I'm looking forward to make SlackBuilds for versions 1.6.X , 1.8.X and Asterisk SCF. I'll publish them as soon as they are ready. -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] how to read mp3
On Tuesday 18 Jan 2011, salaheddine elharit wrote: yes i want to know how can i do in order to read this files using apche Either make a symbolic link to the location of the files from somewhere Apache knows about, using something like # ln -s /path/to/files /path/to/webroot/mp3files/ and set its ownership using # chown -h user:group filename (Apache will only follow links if both ends are owned by the same user) or (possibly better) write a simple script which will read the file from its own location and display it on STDOUT. If doing this you will need at least a content-type: header and maybe a content-length: header (you're probably going to need this anyway in order to be sure you read in the whole file); then a single blank line with just a \n; then the actual contents of the file itself. -- AJS Answers come *after* questions. It's really not hard. -- _ -- 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] Top Posting
On Tue, Jan 18, 2011 at 8:18 AM, Andrew Thomas a...@datavox.co.uk wrote: Why do I top post? Simple. I read every message in the thread - and if there are 10 messages (for example) in that thread - then why should I have to read them all over again on the last one? That's not the alternative (having ten messages above the reply). See this message for an example. I suspect you won't have to scroll at all or read any of the 10+ previous messages. Top posting is here - to stay! It may be. But it would be nice if people cut out the $#@! that is irrelevant to their reply regardless, and were open to hearing what others had to say, rather than saying, I do it this way, it's the best. I also agree this is a pointless discussion because, clearly, nobody is willing to budge, and it has nothing to do with Asterisk. -- _ -- 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] Multiple Registrations
Hi I am researching if there is a practical number of SIP accounts that Asterisk can register against as a UA. I have an idea for a project but it would need to register multiple accounts from multiple providers to work. Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- 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] Top Posting
I also agree this is a pointless discussion because, clearly, nobody is willing to budge, and it has nothing to do with Asterisk. Amen :) [oh no, a bottom post] If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Top Posting
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas Sent: Tuesday, January 18, 2011 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting I also agree this is a pointless discussion because, clearly, nobody is willing to budge, and it has nothing to do with Asterisk. Amen :) It may yet have a point - another few hundred (thousand) of these and the board will blacklist items with the words top post and bottom post :) -- _ -- 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] Top Posting
I also agree this is a pointless discussion because, clearly, nobody is willing to budge, and it has nothing to do with Asterisk. Amen :) It may yet have a point - another few hundred (thousand) of these and the board will blacklist items with the words top post and bottom post :) And maybe If you have received this communication in error... :) -- _ -- 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] Top Posting
I'm top-posting this simply to be consistent with the previous couple posts. I agree that top-posting is preferable for the reason that Andrew pointed out and I prefer no trimming (other than signatures--especially legal disclaimers, etc.) so I can delete every message except the most recent and maintain the entire thread. However, as pointed out a couple days ago, this list's rules specify that we'll respond after the text being responded to, so that's what I'll be doing. PLONK is retro--like bottom-posting :) --Don On Behalf Of Vince Vielhaber Sent: Tuesday, January 18, 2011 9:29 AM I'm top posting this so you will see it and if you don't understand it, look it up. PLONK!! On Tue, 18 Jan 2011, Andrew Thomas wrote: Why do I top post? Simple. I read every message in the thread - and if there are 10 messages (for example) in that thread - then why should I have to read them all over again on the last one? Top posting is here - to stay! Stop being so anal and 'retro'. Bottom posting belongs in forums - top post belongs in e-mail lists. There - said it! -- _ -- 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] Top Posting
On Tue, Jan 18, 2011 at 03:18:49PM -, Andrew Thomas wrote: Why do I top post? Simple. I read every message in the thread - and if there are 10 messages (for example) in that thread - then why should I have to read them all over again on the last one? You mean: why should I have to read 10 messages worth of lines just to figure what you're talking about? It is interesting to note that your mailer (MS-Outlook) has very bad support for threading. In fact, it (combined with the MS-Exchange server) does not really bother reproducing the mail headers that are required to keep the proper threading. Which is why you get a big pile of messages and have to resort to keeping everything in the message itself. Top posting is here - to stay! Top posted content has just been cut off :-) not replying to inane posts like mine :P So, you really want this thread to go on forever? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Top Posting
SEE THE BOTTOM :P -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: 18 January 2011 16:18 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Top Posting On Tue, Jan 18, 2011 at 03:18:49PM -, Andrew Thomas wrote: Why do I top post? Simple. I read every message in the thread - and if there are 10 messages (for example) in that thread - then why should I have to read them all over again on the last one? You mean: why should I have to read 10 messages worth of lines just to figure what you're talking about? It is interesting to note that your mailer (MS-Outlook) has very bad support for threading. In fact, it (combined with the MS-Exchange server) does not really bother reproducing the mail headers that are required to keep the proper threading. Which is why you get a big pile of messages and have to resort to keeping everything in the message itself. Top posting is here - to stay! Top posted content has just been cut off :-) not replying to inane posts like mine :P So, you really want this thread to go on forever? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 You mean: why should I have to read 10 messages worth of lines just to figure what you're talking about? Nope! I mean: why should I have to read the SAME 10 messages worth of lines over and over... It is interesting to note that your mailer (MS-Outlook) has very bad support for threading. In fact, it (combined with the MS-Exchange server) does not really bother reproducing the mail headers that are required to keep the proper threading. Oh dear God! You mean I'm using a Micro$oft product(s)? I'll go shoot myself now! Well, after I've shot every other M$ user! Top posted content has just been cut off :-) I chuckled :-) So, you really want this thread to go on forever? Yeah! I'm having bit of a slow CBA day at work... Watch out - here comes that damned disclaimer again: If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] AST-2011-001: Stack buffer overflow in SIP channel driver
Asterisk Project Security Advisory - AST-2011-001
ProductAsterisk
SummaryStack buffer overflow in SIP channel driver
Nature of Advisory Exploitable Stack Buffer Overflow
SusceptibilityRemote Authenticated Sessions
Severity Moderate
Exploits KnownNo
Reported On January 11, 2011
Reported By Matthew Nicholson
Posted On January 18, 2011
Last Updated OnJanuary 18, 2011
Advisory Contact Matthew Nicholson mnichol...@digium.com
CVE Name
Description When forming an outgoing SIP request while in pedantic mode, a
stack buffer can be made to overflow if supplied with
carefully crafted caller ID information. This vulnerability
also affects the URIENCODE dialplan function and in some
versions of asterisk, the AGI dialplan application as well.
The ast_uri_encode function does not properly respect the size
of its output buffer and can write past the end of it when
encoding URIs.
Resolution The size of the output buffer passed to the ast_uri_encode
function is now properly respected.
In asterisk versions not containing the fix for this issue,
limiting strings originating from remote sources that will be
URI encoded to a length of 40 characters will protect against
this vulnerability.
exten = s,1,Set(CALLERID(num)=${CALLERID(num):0:40})
exten = s,n,Set(CALLERID(name)=${CALLERID(name):0:40})
exten = s,n,Dial(SIP/channel)
The CALLERID(num) and CALLERID(name) channel values, and any
strings passed to the URIENCODE dialplan function should be
limited in this manner.
Affected Versions
Product Release Series
Asterisk Open Source1.2.x All versions
Asterisk Open Source1.4.x All versions
Asterisk Open Source1.6.x All versions
Asterisk Open Source1.8.x All versions
Asterisk Business Edition C.x.x All versions
AsteriskNOW 1.5 All versions
s800i (Asterisk Appliance) 1.2.x All versions
Corrected In
Product Release
Asterisk Open Source 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1,
1.6.2.16.1, 1.8.1.2, 1.8.2.1
Asterisk Business Edition C.3.6.2
Patches
URL Branch
http://downloads.asterisk.org/pub/security/AST-2011-001-1.4.diff1.4
http://downloads.asterisk.org/pub/security/AST-2011-001-1.6.1.diff 1.6.1
http://downloads.asterisk.org/pub/security/AST-2011-001-1.6.2.diff 1.6.2
http://downloads.asterisk.org/pub/security/AST-2011-001-1.8.diff1.8
Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security
This document may be superseded by later versions; if so, the latest
version will be posted at
http://downloads.digium.com/pub/security/AST-2011-001.pdf and
http://downloads.digium.com/pub/security/AST-2011-001.html
Revision History
Date Editor Revisions Made
2011-01-18Matthew NicholsonInitial
[asterisk-users] Asterisk Security Releases: AST-2011-001
The Asterisk Development Team has announced security releases for the following versions of Asterisk: * 1.4.38.1 * 1.4.39.1 * 1.6.1.21 * 1.6.2.15.1 * 1.6.2.16.1 * 1.8.1.2 * 1.8.2.1 These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2, 1.8.1.2, and 1.8.2.1 resolve an issue when forming an outgoing SIP request while in pedantic mode, which can cause a stack buffer to be made to overflow if supplied with carefully crafted caller ID information. The issue and resolution are described in the AST-2011-001 security advisory. For more information about the details of this vulnerability, please read the security advisory AST-2011-001, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.38.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.39.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.21 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.15.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.16.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.1.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.1 Security advisory AST-2011-001 is available at: http://downloads.asterisk.org/pub/security/AST-2011-001.pdf Thank you for your continued support of Asterisk! -- _ -- 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] AST-2011-001: Stack buffer overflow in SIP channel driver
On Tue, 18 Jan 2011, Asterisk Security Team wrote: Asterisk Project Security Advisory - AST-2011-001 ProductAsterisk SummaryStack buffer overflow in SIP channel driver Nature of Advisory Exploitable Stack Buffer Overflow SusceptibilityRemote Authenticated Sessions Severity Moderate Exploits KnownNo Reported On January 11, 2011 Reported By Matthew Nicholson Posted On January 18, 2011 Last Updated OnJanuary 18, 2011 Advisory Contact Matthew Nicholson mnichol...@digium.com CVE Name Description When forming an outgoing SIP request while in pedantic mode, a stack buffer can be made to overflow if supplied with carefully crafted caller ID information. This vulnerability also affects the URIENCODE dialplan function and in some versions of asterisk, the AGI dialplan application as well. The ast_uri_encode function does not properly respect the size of its output buffer and can write past the end of it when encoding URIs. Am I correct in assuming this is only exploitable by registered endpoints? Thanks, 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] AST-2011-001: Stack buffer overflow in SIP channel driver
On 01/18/2011 10:53 AM, Jeff LaCoursiere wrote: On Tue, 18 Jan 2011, Asterisk Security Team wrote: Asterisk Project Security Advisory - AST-2011-001 Product Asterisk Summary Stack buffer overflow in SIP channel driver Nature of Advisory Exploitable Stack Buffer Overflow Susceptibility Remote Authenticated Sessions Severity Moderate Exploits Known No Reported On January 11, 2011 Reported By Matthew Nicholson Posted On January 18, 2011 Last Updated On January 18, 2011 Advisory Contact Matthew Nicholson mnichol...@digium.com CVE Name Description When forming an outgoing SIP request while in pedantic mode, a stack buffer can be made to overflow if supplied with carefully crafted caller ID information. This vulnerability also affects the URIENCODE dialplan function and in some versions of asterisk, the AGI dialplan application as well. The ast_uri_encode function does not properly respect the size of its output buffer and can write past the end of it when encoding URIs. Am I correct in assuming this is only exploitable by registered endpoints? As the advisory says, anyone who can place an authenticated call (if authentication is required) can exploit it. Whether an endpoint is registered or not has nothing to do with whether it can place calls; registration is for delivery of calls to the endpoint. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] Ongoing problem with 1.8
At 01:00 AM 1/18/2011, you wrote: On Tuesday 18 January 2011 01:05:20 Ira wrote: I have tried installing many of the beta versions and most of the release versions of 1.8. I have 3 SIP phones which we use for all our calls. After installing 1.8 the first thing I try is calling out port one of my Digium TDM04 back into port 2. I can see that the call comes in and tries to call all three SIP phones but the phones never ring. Eventually the call goes to voice mail and these error messages pop up. I've read doc/sip-retransmit.txt and as far as I can tell, there's nothing there for me to try. Is there anything else I might try or do to help troubleshoot this. Try running a tcpdump for udp port 5060 while this is occurring. Also, what type of SIP phones are you using? Aastra 480i-CT phones. Is tcpdump port 5060 the syntax you'd like me to use? And I may have neglected to point out, the same system has been running since 1.2.11 or so with basically no issues. Ira -- _ -- 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] how to read mp3
Thank you so much for your response I will try this operation and I will update you as soon as I have any result 2011/1/18 A J Stiles asterisk_l...@earthshod.co.uk On Tuesday 18 Jan 2011, salaheddine elharit wrote: yes i want to know how can i do in order to read this files using apche Either make a symbolic link to the location of the files from somewhere Apache knows about, using something like # ln -s /path/to/files /path/to/webroot/mp3files/ and set its ownership using # chown -h user:group filename (Apache will only follow links if both ends are owned by the same user) or (possibly better) write a simple script which will read the file from its own location and display it on STDOUT. If doing this you will need at least a content-type: header and maybe a content-length: header (you're probably going to need this anyway in order to be sure you read in the whole file); then a single blank line with just a \n; then the actual contents of the file itself. -- AJS Answers come *after* questions. It's really not hard. -- _ -- 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
[asterisk-users] SIP Originate on 1.8.1.1
I am having a problem trying to use originate from the CLI on Asterisk 1.8.1.1. The SIP peer is defined correctly and it works if I dial using my IP phone. When I try to dial from the CLI I get this message: [Jan 18 12:00:09] WARNING[3336]: chan_sip.c:19048 handle_response_invite: Received response: Forbidden from 'Anonymous sip:XX@anonymous.invalid;tag=as67d9024d' This same peer works fine in Asterisk 1.6.2.X so I guess something need to be modified for 1.8? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Sendind e-mail with Hylafax
Hi all, I know Hylafax is an application and not Asterisk but I'd like to post a problem found in configuring such application and Asterisk. I am able to reveive fax,but , I can't receive it in e-mail. Although I put my e-mail in /etc/hylifax/Dispatch I can't receive. Anybody know where I must to add something else in order to make it works! Thanks in advanced!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] Calling rules
Hello. I don't know if this is a problem, but I was expecting a different behavior. Users, have to dial 0 to get an external line, and afterwords the number they want to dial (exe 12345). The thing is: 1-If user dial 012345 there is an error and the call isn't made and the error is handle_request_invite: Call from 'XXX' to extension '012345' rejected because extension not found in context 'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then dials 12345 then it works fine. Should the result be the same? Shouldn't asterisk automatically dial 0, wait and then dial the external number? Best regards, Vitor Flausino -- _ -- 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] Sendind e-mail with Hylafax
snip Although I put my e-mail in /etc/hylifax/Dispatch I can't receive. Flavio Roberto Miranda It may be different for your Hylafax version, etc., but you may want your email in /var/spool/hylafax/etc/FaxDispatch And you probably want to post your questions to the Hylafax list http://lists.hylafax.org/cgi-bin/lsg2.cgi --Don -- _ -- 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] Calling rules
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Carlos
Flausino
Sent: Tuesday, January 18, 2011 12:10 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Calling rules
Hello.
I don't know if this is a problem, but I was expecting a different behavior.
Users, have to dial 0 to get an external line, and afterwords the number
they want to dial (exe 12345). The thing is:
1-If user dial 012345 there is an error and the call isn't made and the
error is handle_request_invite: Call from 'XXX' to extension '012345'
rejected because extension not found in context 'DLPN_DialPlanX'.
2-If user dials 0 waits for the signal, and then dials 12345 then it
works fine.
Should the result be the same? Shouldn't asterisk automatically dial 0,
wait and then dial the external number?
Best regards,
Vitor Flausino
My best guess is that it is a dialplan inconsistency. The standard for
outside line dialing is something like this:
- exten = 0.,1,Dial(DAHDI/1,${EXTEN:1})
Where the dialplan chomps the first digit off of the dialed string.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Ongoing problem with 1.8
On Tuesday 18 January 2011 11:31:07 Ira wrote: At 01:00 AM 1/18/2011, you wrote: On Tuesday 18 January 2011 01:05:20 Ira wrote: I have tried installing many of the beta versions and most of the release versions of 1.8. I have 3 SIP phones which we use for all our calls. After installing 1.8 the first thing I try is calling out port one of my Digium TDM04 back into port 2. I can see that the call comes in and tries to call all three SIP phones but the phones never ring. Eventually the call goes to voice mail and these error messages pop up. I've read doc/sip-retransmit.txt and as far as I can tell, there's nothing there for me to try. Is there anything else I might try or do to help troubleshoot this. Try running a tcpdump for udp port 5060 while this is occurring. Also, what type of SIP phones are you using? Aastra 480i-CT phones. Is tcpdump port 5060 the syntax you'd like me to use? Nope, tcpdump 'udp port 5060'. And I may have neglected to point out, the same system has been running since 1.2.11 or so with basically no issues. While that's a useful data point, it's not relevant to the problem. A significant portion of the SIP stack was re-implemented in 1.8, and Polycom phones are on the desktops of nearly every Asterisk developer. Since you aren't using a Polycom, the SIP stack on that device is implemented differently, causing possible incompatibilities. This is why the tcpdump will be helpful: to figure out what is different and why it doesn't work. -- Tilghman -- _ -- 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] Top Posting
On Tuesday 18 Jan 2011, Don Kelly wrote: PLONK is retro--like bottom-posting :) --Don Retro? For those of us who actually know what PLONK means, it's hilarious. The fact that some people *don't* know what it means only makes it doubly so. Now, here is a link that those of us who remember a time before there was such a thing as http, never mind youtube, will understand: http://www.youtube.com/watch?v=R1JXYgwwDeY Justification: Everybody needs a break from all the telephonical stuff every once in a while :-) And now, seriously: Posting answers *before* the question to which they refer breaks the flow of conversation (point -- counterpoint -- point -- counterpoint, and so forth), making it hard to read; and is also downright rude to anybody reading the archives (which is the first thing any clueful person does when they have a question; chances are, you are not the first person to have asked this, and if an acceptable answer is already recorded then you need never even post). And a mail client that makes it easy to top-post, no matter how popular it might have become thanks to a combination of rampant piracy and illegal acts of a convicted monopoly, is still a *badly-designed* mail client. -- AJS Answers come *after* questions. It's really not hard. -- _ -- 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] Calling rules
My dial plan was generated by asterisk GUI, and the line is:
exten = _0,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,)
where trunk_1 is DAHDI/1
Notice the difference between your 0. and my _0.
Is mine correct?
Best regards,
-vcf
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 18, 2011 7:21:15 PM
Subject: Re: [asterisk-users] Calling rules
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Carlos
Flausino
Sent: Tuesday, January 18, 2011 12:10 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Calling rules
Hello.
I don't know if this is a problem, but I was expecting a different behavior.
Users, have to dial 0 to get an external line, and afterwords the number
they want to dial (exe 12345). The thing is:
1-If user dial 012345 there is an error and the call isn't made and the
error is handle_request_invite: Call from 'XXX' to extension '012345'
rejected because extension not found in context 'DLPN_DialPlanX'.
2-If user dials 0 waits for the signal, and then dials 12345 then it
works fine.
Should the result be the same? Shouldn't asterisk automatically dial 0,
wait and then dial the external number?
Best regards,
Vitor Flausino
My best guess is that it is a dialplan inconsistency. The standard for
outside line dialing is something like this:
- exten = 0.,1,Dial(DAHDI/1,${EXTEN:1})
Where the dialplan chomps the first digit off of the dialed string.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Calling rules
Un-top-posting and discarding cruft...
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote:
Users, have to dial 0 to get an external line, and afterwords the
number they want to dial (exe 12345). The thing is:
1-If user dial 012345 there is an error and the call isn't made and
the error is handle_request_invite: Call from 'XXX' to extension
'012345' rejected because extension not found in context
'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then
dials 12345 then it works fine.
Should the result be the same? Shouldn't asterisk automatically dial
0, wait and then dial the external number?
From: Danny Nicholas da...@debsinc.com
My best guess is that it is a dialplan inconsistency. The standard for
outside line dialing is something like this:
- exten = 0.,1,Dial(DAHDI/1,${EXTEN:1})
Where the dialplan chomps the first digit off of the dialed string.
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote:
exten = _0,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,)
Notice the difference between your 0. and my _0.
Is mine correct?
Both are 'wrong.'
I'm guessing Danny just typed that in off the top of his head -- he forgot
the leading underscore in the pattern.
Please read up on pattern matching. In particular, what '_' and '.' mean.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns
http://www.voip-info.org/wiki/view/Asterisk+Extension+Matching
Should get you started.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Top Posting
Now this thread is really starting to annoy me Dieses Video enthlt Content von WMG. Es ist in deinem Land nicht verfgbar. ..for non english speakers, for US eyes only... Markus PS: to make everybody happy i posted Top and Bottom :-) Am 18.01.2011 19:48, schrieb A J Stiles: On Tuesday 18 Jan 2011, Don Kelly wrote: "PLONK" is "retro"--like bottom-posting :) --Don Retro? For those of us who actually know what PLONK means, it's hilarious. The fact that some people *don't* know what it means only makes it doubly so. Now, here is a link that those of us who remember a time before there was such a thing as http, never mind youtube, will understand: http://www.youtube.com/watch?v=R1JXYgwwDeY Justification: Everybody needs a break from all the telephonical stuff every once in a while :-) And now, seriously: Posting answers *before* the question to which they refer breaks the flow of conversation (point -- counterpoint -- point -- counterpoint, and so forth), making it hard to read; and is also downright rude to anybody reading the archives (which is the first thing any clueful person does when they have a question; chances are, you are not the first person to have asked this, and if an acceptable answer is already recorded then you need never even post). And a mail client that makes it easy to top-post, no matter how popular it might have become thanks to a combination of rampant piracy and illegal acts of a convicted monopoly, is still a *badly-designed* mail client. Now this thread is really starting to annoy me Dieses Video enthlt Content von WMG. Es ist in deinem Land nicht verfgbar. ..for non english speakers, for US eyes only... Markus -- _ -- 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] Top Posting
On Tuesday 18 Jan 2011, Don Kelly wrote: PLONK is retro--like bottom-posting :) --Don boun...@lists.digium.com] On Behalf Of A J Stiles Retro? For those of us who actually know what PLONK means, it's hilarious. Now, here is a link http://www.youtube.com/watch?v=R1JXYgwwDeY Posting answers *before* the question to which they refer breaks the flow of conversation It's clear from your response that you have not followed this entire thread--depending solely on the snippets in the message to which you responded. Thanks for illustrating one of my points. I've been working with computers for over 40 years and don't have the foggiest notion how the Green Day--Wake Me Up When September Ends video applies to Top Posting. --Don -- _ -- 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] Top Posting
On 01/18/2011 10:18 AM, Andrew Thomas wrote: Why do I top post? Simple. I read every message in the thread - and if there are 10 messages (for example) in that thread - then why should I have to read them all over again on the last one? OK, this is a stupid thread, nobody is going to be convinced by anything I say, and by replying it, I am just feeding the trolls and prolonging everyone's agony. But I can't resist. This is the fifth or sixth post that makes this improper assumption. Nobody, I mean NOT A SINGLE PERSON, is advocating bottom posting without trimming. The argument is between: 1.) Top Posting - No Trimming 2.) Bottom or interleaved posting WITH TRIMMING. In fact, I'd rather you top post and trim than bottom post and not. That's one thing we can all agree on. Tom [going back to biting my lip] -- _ -- 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] Top Posting
I've been working with computers for over 40 years and don't have the foggiest notion how the Green Day--Wake Me Up When September Ends video applies to Top Posting. It's a reference to the Everlasting September in 1993. AOL added usenet access to its service, unleashing a horde of dirty, no-good n00bs onto the interwebs. And alas, there was much consternation and gnashing of teeth over the new user's lack of netiquette. -- _ -- 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] Top Posting
I've been working with computers for over 40 years and don't have the foggiest notion how the Green Day--Wake Me Up When September Ends video applies to Top Posting. It's a reference to the Everlasting September in 1993. AOL added usenet access to its service, unleashing a horde of dirty, no-good n00bs onto the interwebs. And alas, there was much consternation and gnashing of teeth over the new user's lack of netiquette. Thanks for the explanation. -- _ -- 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] Calling rules
- Original Message -
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 18, 2011 8:06:47 PM
Subject: Re: [asterisk-users] Calling rules
Un-top-posting and discarding cruft...
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote:
Users, have to dial 0 to get an external line, and afterwords the
number they want to dial (exe 12345). The thing is:
1-If user dial 012345 there is an error and the call isn't made
and
the error is handle_request_invite: Call from 'XXX' to extension
'012345' rejected because extension not found in context
'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then
dials 12345 then it works fine.
Should the result be the same? Shouldn't asterisk automatically
dial
0, wait and then dial the external number?
From: Danny Nicholas da...@debsinc.com
My best guess is that it is a dialplan inconsistency. The
standard for
outside line dialing is something like this:
- exten = 0.,1,Dial(DAHDI/1,${EXTEN:1})
Where the dialplan chomps the first digit off of the dialed
string.
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote:
exten =
_0,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,)
Notice the difference between your 0. and my _0.
Is mine correct?
Both are 'wrong.'
I'm guessing Danny just typed that in off the top of his head -- he
forgot
the leading underscore in the pattern.
Please read up on pattern matching. In particular, what '_' and '.'
mean.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns
http://www.voip-info.org/wiki/view/Asterisk+Extension+Matching
Should get you started.
Correcting the line to:
exten =
_0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,)
problem persists...
any other suggestions?
Best regards,
-vcf
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Re: [asterisk-users] Calling rules
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote:
1-If user dial 012345 there is an error and the call isn't made and
the error is handle_request_invite: Call from 'XXX' to extension
'012345' rejected because extension not found in context
'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then
dials 12345 then it works fine.
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote:
Correcting the line to:
exten =
_0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,)
problem persists...
How about some console output for a 'good' call and a 'failed' call. Also,
a 'show dialplan|dialplan show' for the executed context may yield some
clues.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Occasional robotic sound while call in progress
On Mon, 17 Jan 2011 18:01:14 -0500 Michelle Dupuis mdup...@ocg.ca wrote: We have an application that plays a variety of sound files on one leg of a call (generated by a call file). We've been told that the party listening to the audio files intermittantly hears robotic sounding audio (on/off during the same call). Does the person have a robot? If so, I would suggest they either deactivate their robot while listening to the files, or wear headphones that can muffle its sounds. If not... well... I dunno. That's all I can think of. -- _ -- 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] Calling rules
- Original Message -
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 18, 2011 8:54:11 PM
Subject: Re: [asterisk-users] Calling rules
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote:
1-If user dial 012345 there is an error and the call isn't made
and
the error is handle_request_invite: Call from 'XXX' to extension
'012345' rejected because extension not found in context
'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and
then
dials 12345 then it works fine.
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote:
Correcting the line to:
exten =
_0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,)
problem persists...
How about some console output for a 'good' call and a 'failed' call.
Also,
a 'show dialplan|dialplan show' for the executed context may yield
some
clues.
--
Here goes...
asterisk*CLI dialplan show CallingRule_Outbound_Ch1
[ Context 'CallingRule_Outbound_Ch1' created by 'pbx_config' ]
'_0.' = 1.
Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,) [pbx_config]
-= 1 extension (1 priority) in 1 context. =-
Log when dialing 0924343424
== Using SIP RTP CoS mark 5
-- Executing [0924343424@DLPN_DialPlan1:1] Macro(SIP/6005-0002,
trunkdial-failover-0.3,DAHDI/1/,,trunk_1,) in new stack
-- Executing [s@macro-trunkdial-failover-0.3:1] GotoIf(SIP/6005-0002,
0?1-fmsetcid,1) in new stack
-- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf(SIP/6005-0002,
1?1-setgbobname,1) in new stack
-- Goto (macro-trunkdial-failover-0.3,1-setgbobname,1)
-- Executing [1-setgbobname@macro-trunkdial-failover-0.3:1]
Set(SIP/6005-0002, CALLERID(name)=Glintt) in new stack
-- Executing [1-setgbobname@macro-trunkdial-failover-0.3:2]
Goto(SIP/6005-0002, s,3) in new stack
-- Goto (macro-trunkdial-failover-0.3,s,3)
-- Executing [s@macro-trunkdial-failover-0.3:3] Set(SIP/6005-0002,
CALLERID(num)=222355598) in new stack
-- Executing [s@macro-trunkdial-failover-0.3:4] GotoIf(SIP/6005-0002,
1?1-dial,1) in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
-- Executing [1-dial@macro-trunkdial-failover-0.3:1]
Dial(SIP/6005-0002, DAHDI/1/) in new stack
-- Called 1/
-- DAHDI/1-1 answered SIP/6005-0002
-- Hungup 'DAHDI/1-1'
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero
on 'SIP/6005-0002' in macro 'trunkdial-failover-0.3'
== Spawn extension (DLPN_DialPlan1, 0924343424, 1) exited non-zero on
'SIP/6005-0002'
A normal internal call to 2000 is:
== Using SIP RTP CoS mark 5
-- Executing [2000@DLPN_DialPlan1:1] Directory(SIP/6005-000a,
default,default,f) in new stack
== Parsing '/etc/asterisk/voicemail.conf': == Found
== Parsing '/etc/asterisk/users.conf': == Found
-- SIP/6005-000a Playing 'dir-welcome.ulaw' (language 'en')
-- SIP/6005-000a Playing 'dir-pls-enter.ulaw' (language 'en')
== Spawn extension (DLPN_DialPlan1, 2000, 1) exited non-zero on
'SIP/6005-000a'
Hope helps...
Best regards and thanks in advance...
-vcf
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Re: [asterisk-users] res_fax_digium.so crashing
On Tue, Jan 18, 2011 at 9:02 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 01/16/2011 09:18 PM, Jeremy Kister wrote: On 1/16/2011 4:13 PM, Paul Belanger wrote: I don't believe Digium is blind to its users: Users of Free Fax For Asterisk are not entitled to any Digium technical support [1]. I'm not looking for technical support; I'm just looking for a way to report a bug and possibly help debug/resolve it. But as you know, Digium's website gives FFA users no clear to contact them - even to report problems. issues.asterisk.org has no selection for res_fax_digium since it is not bundled with Asterisk. I call that willful blindness. Don't get me wrong, I'm grateful for FFA and Asterisk in general - I have several running 1.8.2 working correctly. Alternatively, you can generating an unoptimized backtrace [2] and posting the results to the mailing list, seeing if any member of the community has also had an issue. I didnt expect anyone on this list to be interested, but I suppose you're right. This weekend, i set up a new system running Asterisk 1.8.2 on Debian 5.0.7 where the benchmark told me to use res_fax_digium-1.8.0_1.2.1-core2_32 (i also tried generic_32 but that crashed as well). Asterisk correctly detects the fax and transfers to the fax context. But moments after ReceiveFax is called, asterisk crashes, with no tif written where I've directed it to. I have several files including backtraces and config files at http://jeremy.kister.net/tmp/fax/ We have determined the cause of this problem; I'm sorry to say it's an incompatibility between releases that wasn't caught before the releases were made. res_fax.c in Asterisk 1.8.2 contains some changes that make it incompatible with res_fax_digium modules compiled for Asterisk 1.8.0 and 1.8.1, which is what is causing this problem. For now, the res_fax_digium 1.2.x modules are only usable with Asterisk 1.8.0 and 1.8.1; there is a res_fax_digium 1.3.0 release currently being tested for release that is compatible with Asterisk 1.8.2 and later. We'll try to get it out as quickly as we can, and also update the download selector on the Digium website to indicate that the 1.2.x modules should not be used with Asterisk 1.8.2. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org If you are swapping out systems in really busy offices that rely on faxing to keep the doors open, do a whole bunch of testing. FFA (I purchased 3 licenses plus the free one) never worked well and Digium could not support it and the one of the guys on the phone should be unemployed or working in a capacity that does not involve interfacing with customers. Until someone can post there fax show stats and the numbers jive, and I mean hundreds of faxes, not just a dozen, and can also say that FFA works reliably in a fax intensive environment, then I wouldn't even consider it. My experience is based on my Asterisk server with a direct cross connect to Level3's cage in Equinix, VA.using T.38. The problem was not apparent until a few days later when people were complaining about not getting faxes. In my testing, I sent a few dozen, sometimes four at a time with no issues. When you have hundreds of people getting hundreds of faxes, a few dozen test faxes is not really a good real world test in that kind of environment. Obviously, everyone was freaking out because their expected faxes were the most important thing in the world, not to downplay it, many were military documentation and contracts for guys deploying to Iraq and elsewhere. I got it fixed with an all nighter, but I took a beating for the problems for not fully testing and monitoring. After that, nobody had faith in the fax solution. Can anyone that is not affiliated with Digium post their stats and reports from users using T.38? Thanks, Steve T -- _ -- 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] res_fax_digium.so crashing
On 01/18/2011 3:21 PM, Steve Totaro wrote: If you are swapping out systems in really busy offices that rely on faxing to keep the doors open, do a whole bunch of testing. I have no experience with Digium's FFA, beyond installing it and receiving a fax or two. So I can't really agree or disagree with your assertions. However, I can say that iFax does sell a paid/supported version of Hylafax that includes t.38 modems. I've never used their version, but I have been a HylaFAX user for years (both with real modems and POTS lines and with Asterisk PRI and IAXModem), and it is an excellent solution. I also have some experience with their HylaFAX client, which is included with the server, and I can say it is very well done. Might be worth the cash for large fax users like you describe. Tom -- _ -- 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] Calling rules
On 01/18/2011 3:20 PM, Vitor Carlos Flausino wrote: == Spawn extension (DLPN_DialPlan1, 0924343424, 1) exited non-zero on 'SIP/6005-0002' Vitor, Can you please clarify whether the 0 should be received by Asterisk and processed internally, or whether it should be passed to the DAHDI channel by asterisk? In other words, which of the following is your situation: 1.) User dials 0X, asterisk sends 0X to the telco. 2.) User dials 0X, asterisk parses 0, strips it, and sends X to the telco. That might narrow it down. Tom -- _ -- 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] Calling rules
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Carlos
Flausino
Sent: Tuesday, January 18, 2011 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calling rules
snip
Correcting the line to:
exten =
_0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,)
problem persists...
any other suggestions?
Best regards,
What does your trunkdial-failover-0.3 look like?
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Re: [asterisk-users] Calling rules
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote: Log when dialing 0924343424 [snip] A normal internal call to 2000 is: [snip] These two calls do not demonstrate your issue: 1-If user dial 012345 there is an error and the call isn't made and the error is handle_request_invite: Call from 'XXX' to extension '012345' rejected because extension not found in context 'DLPN_DialPlanX'. 2-If user dials 0 waits for the signal, and then dials 12345 then it works fine. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] 1.8.2: dahdi-2.4: calls dropping
Here's a call coming in over PSTN to dahdi/4, connected to a local extension dahdi/1: -- Executing [s@incoming-pstn-line:1] Answer(DAHDI/4-1, ) in new stack .. -- Executing [s@incoming-pstn-line:6] Dial(DAHDI/4-1, DAHDI/g0,36) in new stack -- Called g0 -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 answered DAHDI/4-1 -- Native bridging DAHDI/4-1 and DAHDI/1-1 -- Hanging up on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' == Spawn extension (incoming-pstn-line, s, 6) exited non-zero on 'DAHDI/4-1' -- Hanging up on 'DAHDI/4-1' -- Hungup 'DAHDI/4-1' I'm on dadhi/1. After 10-15 minutes, the call just drops. What gives? sean -- _ -- 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] res_fax_digium.so crashing
On Tue, Jan 18, 2011 at 1:21 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: I got it fixed with an all nighter, but I took a beating for the problems for not fully testing and monitoring. After that, nobody had faith in the fax solution. So is FFA working for you now? What did you have to do to fix it (I like to avoid problems and learning from others is one way to avoid them)? -- _ -- 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] 1.8.2: dahdi-2.4: calls dropping
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Tuesday, January 18, 2011 3:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 1.8.2: dahdi-2.4: calls dropping Here's a call coming in over PSTN to dahdi/4, connected to a local extension dahdi/1: -- Executing [s@incoming-pstn-line:1] Answer(DAHDI/4-1, ) in new stack .. -- Executing [s@incoming-pstn-line:6] Dial(DAHDI/4-1, DAHDI/g0,36) in new stack -- Called g0 -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 answered DAHDI/4-1 -- Native bridging DAHDI/4-1 and DAHDI/1-1 -- Hanging up on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' == Spawn extension (incoming-pstn-line, s, 6) exited non-zero on 'DAHDI/4-1' -- Hanging up on 'DAHDI/4-1' -- Hungup 'DAHDI/4-1' I'm on dadhi/1. After 10-15 minutes, the call just drops. What gives? sean Just a WAG - the bridge isn't really happening and you're getting a dial timeout. -- _ -- 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] 1.8.2: dahdi-2.4: calls dropping
On 01/18/2011 04:06 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Tuesday, January 18, 2011 3:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 1.8.2: dahdi-2.4: calls dropping Here's a call coming in over PSTN to dahdi/4, connected to a local extension dahdi/1: -- Executing [s@incoming-pstn-line:1] Answer(DAHDI/4-1, ) in new stack .. -- Executing [s@incoming-pstn-line:6] Dial(DAHDI/4-1, DAHDI/g0,36) in new stack -- Called g0 -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 answered DAHDI/4-1 -- Native bridging DAHDI/4-1 and DAHDI/1-1 -- Hanging up on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' == Spawn extension (incoming-pstn-line, s, 6) exited non-zero on 'DAHDI/4-1' -- Hanging up on 'DAHDI/4-1' -- Hungup 'DAHDI/4-1' I'm on dadhi/1. After 10-15 minutes, the call just drops. What gives? sean Just a WAG - the bridge isn't really happening and you're getting a dial timeout. If you were running trunk...this is a very good guess. The following commit resolved an issue with bridging that's been in trunk for the past few weeks. http://svn.asterisk.org/view/dahdi?view=revisionrevision=9642 Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] chan_sip.c: Failed to parse contact info
Hello! I have just upgraded to asterisk 1.8.2.1 and see some weird messages in log when client tries to register: [2011-01-19 00:52:47] WARNING[25624] chan_sip.c: Failed to parse contact info [2011-01-19 00:52:50] NOTICE[25624] chan_sip.c: Peer '0010101' is now UNREACHABLE! Last qualify: 105 [2011-01-19 00:53:03] VERBOSE[25624] chan_sip.c: -- Registered SIP '0010101' at 78.84.202.65:37891 So far this is only happeing with some Android SIP client software, X-Lite registers normally. Downgraded to the previous version of asterisk and the warning is gone. Any ideas? Thanks Nick -- _ -- 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] Top Posting
Tom Rymes wrote: On 01/18/2011 10:18 AM, Andrew Thomas wrote: Why do I top post? Simple. I read every message in the thread - and if there are 10 messages (for example) in that thread - then why should I have to read them all over again on the last one? OK, this is a stupid thread, nobody is going to be convinced by anything I say, and by replying it, I am just feeding the trolls and prolonging everyone's agony. But I can't resist. This is the fifth or sixth post that makes this improper assumption. Nobody, I mean NOT A SINGLE PERSON, is advocating bottom posting without trimming. The argument is between: 1.) Top Posting - No Trimming 2.) Bottom or interleaved posting WITH TRIMMING. In fact, I'd rather you top post and trim than bottom post and not. That's one thing we can all agree on. Tom [going back to biting my lip] And yet, SOME of the loudest complainers TRIM NOTHING!! Wading through endless list footers to find a reply certainly doesn't advance the point of the list and Asterisk and VOIP - COMMUNICATION JN -- 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] Top Posting
On Tue, 18 Jan 2011 18:17:31 +0200 Tzafrir Cohen tzafrir.co...@xorcom.com wrote: It is interesting to note that your mailer (MS-Outlook) has very bad support for threading. In fact, it (combined with the MS-Exchange server) does not really bother reproducing the mail headers that are required to keep the proper threading. I think you've hit the nail on the head. We need to ban all versions of outlook until microsoft decides to fix it. -- _ -- 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] 1.8.2: dahdi-2.4: calls dropping
On 01/18/2011 05:27 PM, Shaun Ruffell wrote: On 01/18/2011 04:06 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Tuesday, January 18, 2011 3:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 1.8.2: dahdi-2.4: calls dropping Here's a call coming in over PSTN to dahdi/4, connected to a local extension dahdi/1: -- Executing [s@incoming-pstn-line:1] Answer(DAHDI/4-1, ) in new stack .. -- Executing [s@incoming-pstn-line:6] Dial(DAHDI/4-1, DAHDI/g0,36) in new stack -- Called g0 -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 answered DAHDI/4-1 -- Native bridging DAHDI/4-1 and DAHDI/1-1 -- Hanging up on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' == Spawn extension (incoming-pstn-line, s, 6) exited non-zero on 'DAHDI/4-1' -- Hanging up on 'DAHDI/4-1' -- Hungup 'DAHDI/4-1' I'm on dadhi/1. After 10-15 minutes, the call just drops. What gives? sean Just a WAG - the bridge isn't really happening and you're getting a dial timeout. If you were running trunk...this is a very good guess. The following commit resolved an issue with bridging that's been in trunk for the past few weeks. http://svn.asterisk.org/view/dahdi?view=revisionrevision=9642 Cheers, Shaun Wasn't running trunk. It was the 1.8.2 release. Not sure I understand: the dial timeout is 36 seconds. Yet the call doesn't drop for at least 5, probably 10, maybe more minutes. And no audio was muted while the call was up. It was all just fine. sean -- _ -- 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] Occasional robotic sound while call in progress
On 1/18/11 12:01 AM, Michelle Dupuis wrote: We have an application that plays a variety of sound files on one leg of a call (generated by a call file). We've been told that the party listening to the audio files intermittantly hears robotic sounding audio (on/off during the same call). Anyone have ideas on cause? These calls are on an internal network (lots of network bandwidth), and from a server running 99% idle. Hm I have heard/seen these kind of complaints and in my experience they occur with _very_ low amounts of packet loss. The codec gets confused and can't output the proper audio, just a slightly incorrect version of it. Packet loss like this at the start of a call, which could be caused by some form of NAT traversal via a media proxy where media is only sent both ways when audio has been received from both endpoints, is not unheard of. Network bandwidth is not a very good indicator of the quality of your network Make sure you know if there's packet loss on individual links (managed switches FTW), what the jitter is end to end, etc. -- Andreas Sikkema -- _ -- 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] Top Posting
On Jan 18, 2011, at 6:42 PM, Chad Wallace wrote: We need to ban all versions of outlook until microsoft decides to fix it. Amen. Chris -- - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- 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] 1.8.2: dahdi-2.4: calls dropping
On 1/18/11 6:55 PM, sean darcy wrote: On 01/18/2011 05:27 PM, Shaun Ruffell wrote: On 01/18/2011 04:06 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Here's a call coming in over PSTN to dahdi/4, connected to a local extension dahdi/1: -- Executing [s@incoming-pstn-line:1] Answer(DAHDI/4-1, ) in new stack .. -- Executing [s@incoming-pstn-line:6] Dial(DAHDI/4-1, DAHDI/g0,36) in new stack -- Called g0 -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 answered DAHDI/4-1 -- Native bridging DAHDI/4-1 and DAHDI/1-1 -- Hanging up on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' == Spawn extension (incoming-pstn-line, s, 6) exited non-zero on 'DAHDI/4-1' -- Hanging up on 'DAHDI/4-1' -- Hungup 'DAHDI/4-1' I'm on dadhi/1. After 10-15 minutes, the call just drops. What gives? sean Just a WAG - the bridge isn't really happening and you're getting a dial timeout. If you were running trunk...this is a very good guess. The following commit resolved an issue with bridging that's been in trunk for the past few weeks. http://svn.asterisk.org/view/dahdi?view=revisionrevision=9642 Wasn't running trunk. It was the 1.8.2 release. Not sure I understand: the dial timeout is 36 seconds. Yet the call doesn't drop for at least 5, probably 10, maybe more minutes. And no audio was muted while the call was up. It was all just fine. What card are you using to access the PSTN. It's possible there might be some debug flags you can enable to see if the board thinks the FXS port is flashing. Is this a new installation or are you suddenly having this problem on an old installation? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Top Posting
On 11-01-18 07:42 PM, Chad Wallace wrote: We need to ban all versions of outlook until microsoft decides to fix it. Moderation would be another option (personally opinion). Regardless, we should all now be aware of the rules [1] of the mailing lists. All we can do now is hope people respect them. [1] http://www.asterisk.org/community/rules -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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] progressinband, how much extra CPU load?
Hi everyone, We have an Asterisk 1.4.17 user who has problems with sometimes not getting a ring tone on the calling phone. We're considering setting progressinband = yes, but would like to know how much extra CPU load this will require? If anyone can give something even roughly specific (eg 30% increase) that would be great, rather than just lots. Also, are there any ATAs which are known to not work with progressinband = yes? We have Polycom, Linksys and Audiocode. Thanks for any advice, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk extension not found problem...
Hi All,
I am using Asterisk for one of my projects in OpenBTS. I am having the age
old problem of extension not found when try to make
a call from one registered SIP phone to other registered SIP phone (two
mobile phones connected to Asterisk via OpenBTS).
The exact error thrown on Asterisk CLI is
*chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to
extension 2103 rejected because extension not found*
I have provisioned for both the phones in *sip.conf* and
*extensions.conf*under context
* [sip-external]* but I suspect whatever entry given in extensions.conf,
that file is not getting parsed and extensions are not read.
I have tried all the methods suggested by others in the Asterisk User
community but still the problem remains same. If anybody knows the solution
to this
one, please let me know.
--
Abhinav
Copied below is my sip.conf and extensions.conf
===
*extensions.conf*
===
[globals]
;Using this Macro
[macro-dialGSM]
exten = s,1,Dial(SIP/${ARG1})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy(30)
exten = s-CONGESTION,1,Congestion(30)
exten = s-CHANUNAVAIL,1,playback(ss-noservice)
exten = s-CANCEL,1,Hangup
#include extensions.local.conf
[sip-external]
exten = 2101,1,Macro(dialGSM,2101)
exten = 2102,1,Macro(dialGSM,IMSI310410270465840)
exten = 2103,1,Macro(dialGSM,IMSI404864430002302)
; check for local extensions first
include = sip-local
===
*sip.conf*
==
[general]
; Comment these out if no backhaul is available.
; Use the pair with the shortest latency.
;register = kestrel0:v01pt...@sip.ca1.link2voip.com:5060
;register = kestrel0:v01pt...@sip.ca2.link2voip.com:5060
;register = kestrel0:v01pt...@sip.us1.link2voip.com:5060
;register = kestrel0:v01pt...@sip.us2.link2voip.com:5060
;register = kestrel0:v01pt...@sip.nl1.link2voip.com:5060
;register = kestrel0:v01pt...@sip.nl2.link2voip.com:5060
rtpstart=16386
rtpend=16482
relaxdtmf=yes
[softPhone]
callerid=2101
canreinvite=no
type=friend
context=sip-external
allow=ulaw
allow=gsm
host=dynamic
; provisioned Thu Dec 13 17:15:10 2010
[IMSI310410270465840] ; ATnT SIM card IMSI
callerid=2102
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
dtmfmode=info
; provisioned Thu Dec 14 12:15:10 2010
[IMSI404864430002302] ; Vodafone SIM card IMSI
callerid=2103
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
dtmfmode=info
==
--
_
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Re: [asterisk-users] Asterisk extension not found problem...
On 11-01-18 08:52 PM, abhinav anand wrote: I have tried all the methods suggested by others in the Asterisk User community but still the problem remains same. If anybody knows the solution to this one, please let me know. Which context is your incoming calls using? When you know that, you can run: *CLI dialplan show 2103@incoming context to see if the dialplan actual exists. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Top Posting
Paul Belanger wrote: Moderation would be another option (personally opinion). Regardless, we should all now be aware of the rules [1] of the mailing lists. All we can do now is hope people respect them. [1] http://www.asterisk.org/community/rules -- Paul Belanger Digium, Inc. | Software Developer With that type of trimming and my own trimming, bottom posting works for me, as well as top posting. There is little difference. But with 5 screens of text, , 7-10 repeated messages multiple signature lines and other tripe, bottom posting is a PITA. So if others trim, I am happy to bottom post. Cary Fitch -- _ -- 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] Make ConfBridge hang up on last participant
Is there a way to make ConfBridge hang up on the final participant in a conference (obviously after some sort of initial grace period)? Background - I have just moved all of the phones in my house to separate extensions. As a replacement for the POTS-style shared line, I have implemented a barge in feature; any internal extension is able to join the call of any other internal extension by dialing the extension number followed by *. Behind the scenes, I'm using ChannelRedirect and some additional jiggery pokery to pull everyone into a ConfBridge conference. In the vast majority of cases, I'll end up with 2 internal extensions bridged to an external call. But when the 2 internal extensions hang up, there's nothing to prevent the external party from accidentally staying connected to the bridge, tying up the POTS line or racking up per-minute VoIP charges. Any ideas on how to address this situation would be appreciated. Thanks! -- Ian Pilcher arequip...@gmail.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] Asterisk extension not found problem...
On Tue, 18 Jan 2011, abhinav anand wrote: The exact error thrown on Asterisk CLI is chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to extension 2103 rejected because extension not found What context does 'sip show user IMSI310410270465840' show? What does 'dialplan show 2103@context-from-previous-command' show? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 RTP Engine problem in 1.8.2
hi guys, i have a problem with 1.8 branch no matter which release of 1.8 i'm using. i can't make any sip calls, this is the error message i get on each call: [Jan 18 19:02:15] ERROR[1698] rtp_engine.c: No RTP engine was found. Do you have one loaded? [Jan 18 19:02:15] ERROR[1698] chan_sip.c: Got SDP but have no RTP session allocated. i'm sure that the rtp engine is loaded this is the messages i get when loading rtp engine: module load res_rtp_asterisk.so Loaded res_rtp_asterisk.so == Registered RTP engine 'asterisk' == Parsing '/etc/asterisk/rtp.conf': == Found == RTP Allocating from port range 1650 - 4650 Loaded res_rtp_asterisk.so = (Asterisk RTP Stack) any advice to get rid of this problem? thanks all paradise -- _ -- 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] Top Posting
Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
--
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