[asterisk-users] Asterisk Security: Allow only one phone per sip registration
Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those accounts. 200 being one very sensitive individual. Lets say, an insider, get a new phone or perhaps an xlite and configure it with the same extension, 200. Asterisk will register it as 200 to the new IP address. Now extension 202 call 200. The hacker answers it and pretend is the same person. Do what he want to do and thats it. Question; How can i stop this type of threat Regads Peter -- _ -- 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 Security: Allow only one phone per sip registration
- Original Message - From: Sam Muro resea...@businesstz.com To: asterisk-users@lists.digium.com Sent: Friday, October 14, 2011 2:02:01 AM Subject: [asterisk-users] Asterisk Security: Allow only one phone per sip registration Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those accounts. 200 being one very sensitive individual. Lets say, an insider, get a new phone or perhaps an xlite and configure it with the same extension, 200. Asterisk will register it as 200 to the new IP address. Now extension 202 call 200. The hacker answers it and pretend is the same person. Do what he want to do and thats it. Question; How can i stop this type of threat I would recommend actually setting a different secret field in sip.conf for each device so that your would-be attacker isn't able to register as someone else. Or you could buy a gun. I bet the insider would be very afraid of the gun and would therefore avoid any shenanigans while you were around. This would especially be true if you randomly shot items like coffee cups and plants whenever you thought they were looking at you funny. That'll show 'em. -- _ -- 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 Security: Allow only one phone per sip registration
Terry Wilson wrote: - Original Message - From: Sam Muro resea...@businesstz.com To: asterisk-users@lists.digium.com Sent: Friday, October 14, 2011 2:02:01 AM Subject: [asterisk-users] Asterisk Security: Allow only one phone per sip registration Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those accounts. 200 being one very sensitive individual. Lets say, an insider, get a new phone or perhaps an xlite and configure it with the same extension, 200. Asterisk will register it as 200 to the new IP address. Now extension 202 call 200. The hacker answers it and pretend is the same person. Do what he want to do and thats it. Question; How can i stop this type of threat I would recommend actually setting a different secret field in sip.conf for each device so that your would-be attacker isn't able to register as someone else. Is there a way one can bind sip account to specific mac-address (assume on the same subnet). In this way, even if you know the username/secret, you will still have to use the same physical phone, unless you play with mac-address. Or you could buy a gun. I bet the insider would be very afraid of the gun and would therefore avoid any shenanigans while you were around. This would especially be true if you randomly shot items like coffee cups and plants whenever you thought they were looking at you funny. That'll show 'em. Lol! Here they will name you a terrorist -- _ -- 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] Asterisk Security: Allow only one phone per sip registration
On Fri, 2011-10-14 at 10:02 +0300, Muro, Sam wrote: Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those accounts. 200 being one very sensitive individual. Lets say, an insider, get a new phone or perhaps an xlite and configure it with the same extension, 200. Asterisk will register it as 200 to the new IP address. Now extension 202 call 200. The hacker answers it and pretend is the same person. Do what he want to do and thats it. Question; How can i stop this type of threat Regads Peter Perhaps use secrets? afaicr the secrets you have to provide for hardphone and softphone are readonly. If you avoid something like secret or welcome or the involved hostname, but instead use a 15 char long generated pwd, he'll have a long time trying all the possibilities And different pwds for each phone. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration
Is there a way one can bind sip account to specific mac-address (assume on the same subnet). In this way, even if you know the username/secret, you will still have to use the same physical phone, unless you play with mac-address. No. And mac addresses are easily spoofed so it would not help. Use passwords. Keep them safe. -- _ -- 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 Security: Allow only one phone per sip registration
Terry Wilson wrote: Is there a way one can bind sip account to specific mac-address (assume on the same subnet). In this way, even if you know the username/secret, you will still have to use the same physical phone, unless you play with mac-address. No. And mac addresses are easily spoofed so it would not help. Use passwords. Keep them safe. Thanks. Let me see how best i can complicate them per phone. Ooops, 1000 sip phones -- _ -- 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] Asterisk Security: Allow only one phone per sip registration
Thanks. Let me see how best i can complicate them per phone. Ooops, 1000 sip phones If it were me, I would look into Asterisk Realtime for handling the SIP phones. I would then write a script to generate the configs for the phones into the SIP realtime database with random passwords. Match up the phones with the accounts and provision the phones. You would most likely use a provisioning server of some kind to generate the actual phone configurations. You can check out the res_phoneprov module in Asterisk, find another one somewhere, or write your own. Many people tend to write their own for large installations. I did. If you have a big installation like this and are wondering about things like whether mac addresses should be used for security, it might also be a good idea to hire a consultant. Check out the asterisk-biz mailing list. Terry -- _ -- 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] Get the total amount of lines/channels for a SIP-trunk?
Hello! Is it possible for Asterisk to get the total amount of phone lines/channels on a SIP-trunk? Is there some kind of SIP-request to the provider or do I have to call the trunk provider and ask every time? (I want to know the total amount available on the trunk, so sip show channels won't help me out here) Best regards Tobias Steen -- _ -- 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 Security: Allow only one phone per sip registration
Thanks Terry! Let me think of all possibilities and shall holla. Can you be one? Terry Wilson wrote: Thanks. Let me see how best i can complicate them per phone. Ooops, 1000 sip phones If it were me, I would look into Asterisk Realtime for handling the SIP phones. I would then write a script to generate the configs for the phones into the SIP realtime database with random passwords. Match up the phones with the accounts and provision the phones. You would most likely use a provisioning server of some kind to generate the actual phone configurations. You can check out the res_phoneprov module in Asterisk, find another one somewhere, or write your own. Many people tend to write their own for large installations. I did. If you have a big installation like this and are wondering about things like whether mac addresses should be used for security, it might also be a good idea to hire a consultant. Check out the asterisk-biz mailing list. Terry -- _ -- 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] one way voice with IVR
Hi all, I'm stuck on a tricky problem. I set up an Asterisk 1.4.26.2 on a box with a bunch of Snom Phones. When I call an IVR I get the damned one way voice phenomena. It is not randomic, it happens all the time. I tried to upgrade the snom firmware to 7.3.30 but nothing changed. If I call a phone I get a normal conversation and no problem occurs if I (blind) transfer the call. If I use a IAX phone everything is fine. I think it is a SIP problem but I checked the sip files and they seem ok. Tones seems to pass since the caller (me) can make a choice from within the IVR menu. Sincerely, I haven't any idea left to try... Any hints? Thanks Giorgio -- _ -- 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 Security: Allow only one phone per sip registration
On Friday 14 October 2011, Muro, Sam wrote: Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those accounts. 200 being one very sensitive individual. Lets say, an insider, get a new phone or perhaps an xlite and configure it with the same extension, 200. Asterisk will register it as 200 to the new IP address. Now extension 202 call 200. The hacker answers it and pretend is the same person. Do what he want to do and thats it. Question; How can i stop this type of threat Be careful who you employ and how you treat them :) Once someone has physical access to your equipment, all bets are off . -- AJS Answers come *after* questions. -- _ -- 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 Security: Allow only one phone per sip registration
the best way to handle large sip client base is using provisioning interface. Even though you can create configuration files and server them with asterisk+extensions, you need to consider security aspects of this approach as well. Using tftp or simple protocols to server config files works on LAN, but does not scale for large installs (my opinion). HTTP is a better choice, but then all the information is passed in clear. HTTPS is obviously a better choice with SSL, but if your devices can't handle SSL it will become a problem. A good solution is to provide a mix depending on your SIP client capabilities. In the configuration you can supply password/secret as other recommend and any other device specific configuration (i.e. preferred codec, DNS, etc). it really becomes a powerful tool. You also need to have a management capabilities to generate and update your configuration profile either for individual devices (i.e. changes users's secret) or in bulk (change DNS servers or proxy on 1000 SIP clients at once). SIP clients will also need to have capabilities to poll for this configuration on reboot or on regular poll intervals. If you are doing that on the poll interval, don't make it the interval too short (i.e. minutes). I would say 3-4 times a day is a good starting point. If your network is pretty static and not much information changes you can even make it 1-2 a day and experiment with your network load. On Oct 14, 2011, at 7:09 AM, A J Stiles wrote: On Friday 14 October 2011, Muro, Sam wrote: Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those accounts. 200 being one very sensitive individual. Lets say, an insider, get a new phone or perhaps an xlite and configure it with the same extension, 200. Asterisk will register it as 200 to the new IP address. Now extension 202 call 200. The hacker answers it and pretend is the same person. Do what he want to do and thats it. Question; How can i stop this type of threat Be careful who you employ and how you treat them :) Once someone has physical access to your equipment, all bets are off . -- AJS Answers come *after* questions. -- _ -- 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] Asterisk Security: Allow only one phone per sip registration
Thanks A.J I know and I can assure you no one will get that physical access to the system. A J Stiles wrote: On Friday 14 October 2011, Muro, Sam wrote: Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those accounts. 200 being one very sensitive individual. Lets say, an insider, get a new phone or perhaps an xlite and configure it with the same extension, 200. Asterisk will register it as 200 to the new IP address. Now extension 202 call 200. The hacker answers it and pretend is the same person. Do what he want to do and thats it. Question; How can i stop this type of threat Be careful who you employ and how you treat them :) Once someone has physical access to your equipment, all bets are off . -- AJS Answers come *after* questions. -- _ -- 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] one way voice with IVR
Hi Giorgio, This behavior usually indicates some sort of firewall issue where either inbound or outbound rtp traffic (the voice) is being blocked or not routed correctly, though the SIP traffic makes it through (as the call is being set up correctly). This could also be multiple SIP extensions attempting to register over the same port from a single location. What kind of firewall/router is being used at the location where these Snoms are registering from? Are all the phones attempting to register over port 5060 or are you setting them up to register over unique ports to Asterisk? If you are setting them up to register over specific ports, are they registering over those ports according to 'asterisk show peers'? Also, is your asterisk box local or hosted somewhere? Comparing IAX2 to SIP registrations is somewhat different: IAX2 tends to handle cutting through firewalls better though SIP is far better supported by everyone. On Fri, Oct 14, 2011 at 6:21 AM, gincantalupo gincantal...@fgasoftware.comwrote: Hi all, I'm stuck on a tricky problem. I set up an Asterisk 1.4.26.2 on a box with a bunch of Snom Phones. When I call an IVR I get the damned one way voice phenomena. It is not randomic, it happens all the time. I tried to upgrade the snom firmware to 7.3.30 but nothing changed. If I call a phone I get a normal conversation and no problem occurs if I (blind) transfer the call. If I use a IAX phone everything is fine. I think it is a SIP problem but I checked the sip files and they seem ok. Tones seems to pass since the caller (me) can make a choice from within the IVR menu. Sincerely, I haven't any idea left to try... Any hints? Thanks Giorgio -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- * * *John Knight* Classic City Telco LLC *Email:* j...@classiccitytelco.com | *Main:* (706) 995-0200 *Direct:* (706) 995-0201 | *Mobile:* (706) 255-9203 http://www.classiccitytelco.com * * logo.png-- _ -- 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] one way voice with IVR
Netstat -anp has been useful in finding this error for me in the past. A normal Asterisk call will have 2 or 4 udp connections to carry traffic to/from phone to pbx. On a one-way call, this will be an odd count. Then you can check your rtp.conf and firewall and see how the channel got blocked. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Knight Sent: Friday, October 14, 2011 8:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] one way voice with IVR Hi Giorgio, This behavior usually indicates some sort of firewall issue where either inbound or outbound rtp traffic (the voice) is being blocked or not routed correctly, though the SIP traffic makes it through (as the call is being set up correctly). This could also be multiple SIP extensions attempting to register over the same port from a single location. What kind of firewall/router is being used at the location where these Snoms are registering from? Are all the phones attempting to register over port 5060 or are you setting them up to register over unique ports to Asterisk? If you are setting them up to register over specific ports, are they registering over those ports according to 'asterisk show peers'? Also, is your asterisk box local or hosted somewhere? Comparing IAX2 to SIP registrations is somewhat different: IAX2 tends to handle cutting through firewalls better though SIP is far better supported by everyone. On Fri, Oct 14, 2011 at 6:21 AM, gincantalupo gincantal...@fgasoftware.com wrote: Hi all, I'm stuck on a tricky problem. I set up an Asterisk 1.4.26.2 on a box with a bunch of Snom Phones. When I call an IVR I get the damned one way voice phenomena. It is not randomic, it happens all the time. I tried to upgrade the snom firmware to 7.3.30 but nothing changed. If I call a phone I get a normal conversation and no problem occurs if I (blind) transfer the call. If I use a IAX phone everything is fine. I think it is a SIP problem but I checked the sip files and they seem ok. Tones seems to pass since the caller (me) can make a choice from within the IVR menu. Sincerely, I haven't any idea left to try... Any hints? Thanks Giorgio -- _ -- 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 -- John Knight Classic City Telco LLC Email: j...@classiccitytelco.com | Main: (706) 995-0200 Direct: (706) 995-0201 | Mobile: (706) 255-9203 image001.png-- _ -- 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] Get the total amount of lines/channels for a SIP-trunk?
Not the answer you are looking for, but some controlling factors are 1. The available bandwidth. Since a call takes 30-90K depending on the codec unless you are using a compression codec that can reduce this to 5K or so, your number of channels available will be limited by this. 2. Call-limit in sip.conf - if you set this low, you are limiting the number of trunks to that (found out hard way) 3. Your range in rtp.conf - you get one line for every 4 digits of range, IE 10001-10019 is 5 lines. My best guess is that this isn't controlled by Asterisk per se like PRI/DAHDI lines, but is determined by the provider and probably not offered as a query function. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tobias Steen Sent: Friday, October 14, 2011 4:17 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Get the total amount of lines/channels for a SIP-trunk? Hello! Is it possible for Asterisk to get the total amount of phone lines/channels on a SIP-trunk? Is there some kind of SIP-request to the provider or do I have to call the trunk provider and ask every time? (I want to know the total amount available on the trunk, so sip show channels won't help me out here) Best regards Tobias Steen -- _ -- 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] Problem with outbound dialing from remote phone
I have a real head scratcher . . . We have several employees who work from home. All have Polycom 501's that register to our office Asterisk 1.6.x server and communicate using SIP g729a. About two weeks ago, one of these remote users starting experiencing a problem with a previously working phone: a. She could receive inbound calls, b. She can place outbound calls to internal extensions c. She cannot place outbound calls to external destinations. I brought up the Asterisk CLI and had her dial outbound while I watched. The calls to internal extensions are processing as they should. However, I do not see the external dial attempts ever getting to the server. This is odd because there is absolutely nothing in the programming of the phone that distinguishes one from the other. I had her key in several strings on nonsense and I saw some, but not all of them. So, I programmed another phone, and tested it thoroughly from my own remote location. Phone works fine inbound and outbound. I then shipped the phone to the user. User received new phone, plugged it in. It registers to the Asterisk server just fine. It receives inbound calls, however this one cannot dial out at all. I see no dial attempts whatsoever on CLI. If she plugs the old phone back in, she can still dial internal extensions. I know the problem is not with the phone dial pattern, as I've had her key in the number and then press the Dial key. Besides, the phone worked 24 hours earlier from a different location. The sip.conf configuration has not changed from when the phone worked properly: [1234] type=friend regext=1234 context=longdistance secret=* callerid=User Name 1234 host=dynamic qualify=yes mailbox=1234 permit=0.0.0.0/0.0.0.0 I've checked all log files, and for the failed attempts I see nothing ever getting to the server. I don't think the problem is with the phone. Any ideas, suggestions, etc., would be greatly appreciated. If I need to provide additional info please advise. Thanks. The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
Adam Robins wrote: Any ideas, suggestions, etc., would be greatly appreciated My guess that the Polycom digitmap isn't being loaded (sip.cfg). I'm sure if she were to dial the phone number and then press 'send' soft key, it'd probably dial. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Problem with outbound dialing from remote phone
What happens if she keys in the number+# then presses dial? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problem with outbound dialing from remote phone I have a real head scratcher . . . We have several employees who work from home. All have Polycom 501's that register to our office Asterisk 1.6.x server and communicate using SIP g729a. About two weeks ago, one of these remote users starting experiencing a problem with a previously working phone: a. She could receive inbound calls, b. She can place outbound calls to internal extensions c. She cannot place outbound calls to external destinations. I brought up the Asterisk CLI and had her dial outbound while I watched. The calls to internal extensions are processing as they should. However, I do not see the external dial attempts ever getting to the server. This is odd because there is absolutely nothing in the programming of the phone that distinguishes one from the other. I had her key in several strings on nonsense and I saw some, but not all of them. So, I programmed another phone, and tested it thoroughly from my own remote location. Phone works fine inbound and outbound. I then shipped the phone to the user. User received new phone, plugged it in. It registers to the Asterisk server just fine. It receives inbound calls, however this one cannot dial out at all. I see no dial attempts whatsoever on CLI. If she plugs the old phone back in, she can still dial internal extensions. I know the problem is not with the phone dial pattern, as I've had her key in the number and then press the Dial key. Besides, the phone worked 24 hours earlier from a different location. The sip.conf configuration has not changed from when the phone worked properly: [1234] type=friend regext=1234 context=longdistance secret=* callerid=User Name 1234 host=dynamic qualify=yes mailbox=1234 permit=0.0.0.0/0.0.0.0 I've checked all log files, and for the failed attempts I see nothing ever getting to the server. I don't think the problem is with the phone. Any ideas, suggestions, etc., would be greatly appreciated. If I need to provide additional info please advise. Thanks. The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
I wish it was that easy. That is one of the first things we tried. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Adam Robins wrote: Any ideas, suggestions, etc., would be greatly appreciated My guess that the Polycom digitmap isn't being loaded (sip.cfg). I'm sure if she were to dial the phone number and then press 'send' soft key, it'd probably dial. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
No change, thanks -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 14, 2011 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone What happens if she keys in the number+# then presses dial? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problem with outbound dialing from remote phone I have a real head scratcher . . . We have several employees who work from home. All have Polycom 501's that register to our office Asterisk 1.6.x server and communicate using SIP g729a. About two weeks ago, one of these remote users starting experiencing a problem with a previously working phone: a. She could receive inbound calls, b. She can place outbound calls to internal extensions c. She cannot place outbound calls to external destinations. I brought up the Asterisk CLI and had her dial outbound while I watched. The calls to internal extensions are processing as they should. However, I do not see the external dial attempts ever getting to the server. This is odd because there is absolutely nothing in the programming of the phone that distinguishes one from the other. I had her key in several strings on nonsense and I saw some, but not all of them. So, I programmed another phone, and tested it thoroughly from my own remote location. Phone works fine inbound and outbound. I then shipped the phone to the user. User received new phone, plugged it in. It registers to the Asterisk server just fine. It receives inbound calls, however this one cannot dial out at all. I see no dial attempts whatsoever on CLI. If she plugs the old phone back in, she can still dial internal extensions. I know the problem is not with the phone dial pattern, as I've had her key in the number and then press the Dial key. Besides, the phone worked 24 hours earlier from a different location. The sip.conf configuration has not changed from when the phone worked properly: [1234] type=friend regext=1234 context=longdistance secret=* callerid=User Name 1234 host=dynamic qualify=yes mailbox=1234 permit=0.0.0.0/0.0.0.0 I've checked all log files, and for the failed attempts I see nothing ever getting to the server. I don't think the problem is with the phone. Any ideas, suggestions, etc., would be greatly appreciated. If I need to provide additional info please advise. Thanks. The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
Adam Robins wrote: No change, thanks Well, In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Problem with outbound dialing from remote phone
I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Adam Robins wrote: No change, thanks Well, In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
I am assuming you are using a provisioning server. If the phone is running firmware 3.2 or earlier you can access the phone web interface and confirm the dialplan active on the phone is the same as what you set in the config file on the server. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Adam Robins wrote: No change, thanks Well, In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
The phone was originally provisioned from an FTP server when it was inside our network. Once in the field, the phone no longer has access to that server (it could if I wanted it to). It boots using the last known config, which worked before shipping. I've been doing it this way for 5+ years. This is the first problem of its kind.I can get into the phone by RDPing to the users laptop over VPN and then accessing the phone web interface. I will try that. Please remember, I've already tried two phones, both of which worked fine at another remote location prior to shipping, having been programmed from good config files. The first one actually worked fine at this remote location for a period of time and then suddenly went bad. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, October 14, 2011 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I am assuming you are using a provisioning server. If the phone is running firmware 3.2 or earlier you can access the phone web interface and confirm the dialplan active on the phone is the same as what you set in the config file on the server. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Adam Robins wrote: No change, thanks Well, In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
Hey, Can you enable sip trace for that particular sip extension. This sounds weird that while other INVITES from the phone are reaching but the external extensions are filtered. If there are no invites for external calls only then more chances are that the phone is using some dial pattern(phonebook help) etc like Doug and Eric said. Sometimes in asterisk console I don't see anything in logs if the Sip extensions' context don't contain the number that is being dialled Do you've access to any phone debugging console? Sounds like problem is somewhere around She :p j/k . -- Regards, Sammy. On Fri, Oct 14, 2011 at 10:34 PM, Adam Robins arob...@pharmacentra.comwrote: The phone was originally provisioned from an FTP server when it was inside our network. Once in the field, the phone no longer has access to that server (it could if I wanted it to). It boots using the last known config, which worked before shipping. I've been doing it this way for 5+ years. This is the first problem of its kind.I can get into the phone by RDPing to the users laptop over VPN and then accessing the phone web interface. I will try that. Please remember, I've already tried two phones, both of which worked fine at another remote location prior to shipping, having been programmed from good config files. The first one actually worked fine at this remote location for a period of time and then suddenly went bad. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, October 14, 2011 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I am assuming you are using a provisioning server. If the phone is running firmware 3.2 or earlier you can access the phone web interface and confirm the dialplan active on the phone is the same as what you set in the config file on the server. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Adam Robins wrote: No change, thanks Well, In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified
Re: [asterisk-users] Problem with outbound dialing from remote phone
Turned on sip set debug peer 1234. I see the qualify messages. I see when she calls me on my internal extension. I see no SIP messages at all when she calls my cell phone. I understand what Doug and Eric are saying. I need to get into the phone's web interface to see how it is programmed just to validate that the phone is still as I programmed it. What is strange is: a. Phone A can dial local extensions but not external, so I send her Phone B. b. Phone B cant dial outbound at all c. Both phones were successfully tested for both call types prior to shipping and were not in any way reconfigured subsequent to testing. d. I have not modified the digitmap is sip.cfg in years, and even so, entering the number and then pressing 'Dial' doesn't work either. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind Sent: Friday, October 14, 2011 2:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Hey, Can you enable sip trace for that particular sip extension. This sounds weird that while other INVITES from the phone are reaching but the external extensions are filtered. If there are no invites for external calls only then more chances are that the phone is using some dial pattern(phonebook help) etc like Doug and Eric said. Sometimes in asterisk console I don't see anything in logs if the Sip extensions' context don't contain the number that is being dialled Do you've access to any phone debugging console? Sounds like problem is somewhere around She :p j/k . -- Regards, Sammy. On Fri, Oct 14, 2011 at 10:34 PM, Adam Robins arob...@pharmacentra.commailto:arob...@pharmacentra.com wrote: The phone was originally provisioned from an FTP server when it was inside our network. Once in the field, the phone no longer has access to that server (it could if I wanted it to). It boots using the last known config, which worked before shipping. I've been doing it this way for 5+ years. This is the first problem of its kind.I can get into the phone by RDPing to the users laptop over VPN and then accessing the phone web interface. I will try that. Please remember, I've already tried two phones, both of which worked fine at another remote location prior to shipping, having been programmed from good config files. The first one actually worked fine at this remote location for a period of time and then suddenly went bad. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, October 14, 2011 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I am assuming you are using a provisioning server. If the phone is running firmware 3.2 or earlier you can access the phone web interface and confirm the dialplan active on the phone is the same as what you set in the config file on the server. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Adam Robins wrote: No change, thanks Well, In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 The information contained in this transmission may contain privileged and confidential
Re: [asterisk-users] Problem with outbound dialing from remote phone
I use 501's here and I can pull up the settings by typing http://1.2.3.4/index.htm - where 1.2.3.4 http://1.2.3.4/index.htm%20-%20where%201.2.3.4 is the IP address of the phone. If you can do that, perhaps something there will be of use to you. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Turned on sip set debug peer 1234. I see the qualify messages. I see when she calls me on my internal extension. I see no SIP messages at all when she calls my cell phone. I understand what Doug and Eric are saying. I need to get into the phone's web interface to see how it is programmed just to validate that the phone is still as I programmed it. What is strange is: a. Phone A can dial local extensions but not external, so I send her Phone B. b. Phone B cant dial outbound at all c. Both phones were successfully tested for both call types prior to shipping and were not in any way reconfigured subsequent to testing. d. I have not modified the digitmap is sip.cfg in years, and even so, entering the number and then pressing 'Dial' doesn't work either. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind Sent: Friday, October 14, 2011 2:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Hey, Can you enable sip trace for that particular sip extension. This sounds weird that while other INVITES from the phone are reaching but the external extensions are filtered. If there are no invites for external calls only then more chances are that the phone is using some dial pattern(phonebook help) etc like Doug and Eric said. Sometimes in asterisk console I don't see anything in logs if the Sip extensions' context don't contain the number that is being dialled Do you've access to any phone debugging console? Sounds like problem is somewhere around She :p j/k . -- Regards, Sammy. On Fri, Oct 14, 2011 at 10:34 PM, Adam Robins arob...@pharmacentra.com wrote: The phone was originally provisioned from an FTP server when it was inside our network. Once in the field, the phone no longer has access to that server (it could if I wanted it to). It boots using the last known config, which worked before shipping. I've been doing it this way for 5+ years. This is the first problem of its kind.I can get into the phone by RDPing to the users laptop over VPN and then accessing the phone web interface. I will try that. Please remember, I've already tried two phones, both of which worked fine at another remote location prior to shipping, having been programmed from good config files. The first one actually worked fine at this remote location for a period of time and then suddenly went bad. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, October 14, 2011 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I am assuming you are using a provisioning server. If the phone is running firmware 3.2 or earlier you can access the phone web interface and confirm the dialplan active on the phone is the same as what you set in the config file on the server. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Adam Robins wrote: No change, thanks Well, In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
[asterisk-users] Scheduled Maintenance for Asterisk Project community services
On Thursday, October 15th, 2011,the Asterisk community services listed below will be undergoing maintenance (software upgrades and updates). The services will be shut down at approximately 9:00 PM CDT (2:00 AM October 15 UTC), and will return no later than 10:00 PM CDT. We apologize in advance for any inconvenience this may cause. The affected services are: issues.asterisk.org/jira -- _ -- 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] Problem with outbound dialing from remote phone
Thanks I will do that. The user is remote, so I must first RDP into her home network and do it from her PC. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 14, 2011 3:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I use 501's here and I can pull up the settings by typing http://1.2.3.4/index.htm - where 1.2.3.4http://1.2.3.4/index.htm%20-%20where%201.2.3.4 is the IP address of the phone. If you can do that, perhaps something there will be of use to you. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Turned on sip set debug peer 1234. I see the qualify messages. I see when she calls me on my internal extension. I see no SIP messages at all when she calls my cell phone. I understand what Doug and Eric are saying. I need to get into the phone's web interface to see how it is programmed just to validate that the phone is still as I programmed it. What is strange is: a. Phone A can dial local extensions but not external, so I send her Phone B. b. Phone B cant dial outbound at all c. Both phones were successfully tested for both call types prior to shipping and were not in any way reconfigured subsequent to testing. d. I have not modified the digitmap is sip.cfg in years, and even so, entering the number and then pressing 'Dial' doesn't work either. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind Sent: Friday, October 14, 2011 2:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Hey, Can you enable sip trace for that particular sip extension. This sounds weird that while other INVITES from the phone are reaching but the external extensions are filtered. If there are no invites for external calls only then more chances are that the phone is using some dial pattern(phonebook help) etc like Doug and Eric said. Sometimes in asterisk console I don't see anything in logs if the Sip extensions' context don't contain the number that is being dialled Do you've access to any phone debugging console? Sounds like problem is somewhere around She :p j/k . -- Regards, Sammy. On Fri, Oct 14, 2011 at 10:34 PM, Adam Robins arob...@pharmacentra.commailto:arob...@pharmacentra.com wrote: The phone was originally provisioned from an FTP server when it was inside our network. Once in the field, the phone no longer has access to that server (it could if I wanted it to). It boots using the last known config, which worked before shipping. I've been doing it this way for 5+ years. This is the first problem of its kind.I can get into the phone by RDPing to the users laptop over VPN and then accessing the phone web interface. I will try that. Please remember, I've already tried two phones, both of which worked fine at another remote location prior to shipping, having been programmed from good config files. The first one actually worked fine at this remote location for a period of time and then suddenly went bad. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, October 14, 2011 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I am assuming you are using a provisioning server. If the phone is running firmware 3.2 or earlier you can access the phone web interface and confirm the dialplan active on the phone is the same as what you set in the config file on the server. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From:
[asterisk-users] Correction: Scheduled Maintenance for Asterisk Project community services
This is a correction to the recent scheduled maintenance notice. The maintenance will take place at 9 PM CDT Saturday, October 15th, 2011 (2:00 AM October 16 UTC). -- _ -- 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] Problem with outbound dialing from remote phone
Check for any kind of SIP interference from the end user's router. Thanks, --Warren Selby, dCAP On Oct 14, 2011, at 2:38 PM, Adam Robins arob...@pharmacentra.com wrote: Thanks I will do that. The user is remote, so I must first RDP into her home network and do it from her PC. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 14, 2011 3:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I use 501’s here and I can pull up the settings by typing http://1.2.3.4/index.htm - where 1.2.3.4 is the IP address of the phone. If you can do that, perhaps something there will be of use to you. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Turned on “sip set debug peer 1234”. I see the qualify messages. I see when she calls me on my internal extension. I see no SIP messages at all when she calls my cell phone. I understand what Doug and Eric are saying. I need to get into the phone’s web interface to see how it is programmed just to validate that the phone is still as I programmed it. What is strange is: a. Phone “A” can dial local extensions but not external, so I send her Phone “B”. b. Phone “B” cant dial outbound at all c. Both phones were successfully tested for both call types prior to shipping and were not in any way reconfigured subsequent to testing. d. I have not modified the digitmap is sip.cfg in years, and even so, entering the number and then pressing ‘Dial’ doesn’t work either. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind Sent: Friday, October 14, 2011 2:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Hey, Can you enable sip trace for that particular sip extension. This sounds weird that while other INVITES from the phone are reaching but the external extensions are filtered. If there are no invites for external calls only then more chances are that the phone is using some dial pattern(phonebook help) etc like Doug and Eric said. Sometimes in asterisk console I don't see anything in logs if the Sip extensions' context don't contain the number that is being dialled Do you've access to any phone debugging console? Sounds like problem is somewhere around She :p j/k . -- Regards, Sammy. On Fri, Oct 14, 2011 at 10:34 PM, Adam Robins arob...@pharmacentra.com wrote: The phone was originally provisioned from an FTP server when it was inside our network. Once in the field, the phone no longer has access to that server (it could if I wanted it to). It boots using the last known config, which worked before shipping. I've been doing it this way for 5+ years. This is the first problem of its kind.I can get into the phone by RDPing to the users laptop over VPN and then accessing the phone web interface. I will try that. Please remember, I've already tried two phones, both of which worked fine at another remote location prior to shipping, having been programmed from good config files. The first one actually worked fine at this remote location for a period of time and then suddenly went bad. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, October 14, 2011 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I am assuming you are using a provisioning server. If the phone is running firmware 3.2 or earlier you can access the phone web interface and confirm the dialplan active on the phone is the same as what you set in the config file on the server. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 12:48 PM To: Asterisk Users Mailing List - Non-Commercial
Re: [asterisk-users] Problem with outbound dialing from remote phone
Un-top-posting and trimming cruft... On Fri, 14 Oct 2011, Adam Robins wrote: Thanks I will do that. The user is remote, so I must first RDP into her home network and do it from her PC. Since you have access to the Asterisk server's command line, you could use wget to retrieve the index page from the phone outputting to a file and scp the file back to you for your local viewing pleasure. -- 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
Re: [asterisk-users] Free ticket for Astricon
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 2011-10-13 17:04, Sylvain Boily a écrit : Hello everybody, I've got a ticket for Astricon but i can't go ... So i don't want to lost it for nothing and i want to give it for free to the asterisk community. Just send me a tweet to @avencall with astricon in the tweet. I will choose at random the tweet for the winner . The end is friday afternoon at 4am. Have fun. Sylvain Hello, game is finish, Charles is the winner. Have fun at Astricon :) Hope to coming next year :( - -- Sylvain BOILY Proformatique Inc - 2590, boul. Laurier, local 770, Québec, G1V 4M6 Tel. : +1 418 476 5458 - Fax. : +33 1 41 38 99 70 Email : sbo...@proformatique.com - http://proformatique.com/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.11 (GNU/Linux) iQEcBAEBAgAGBQJOmJ0cAAoJEFfSa6I0jU6KmxQIAKqit9BLyZFuLmYqpj6PL2VM /siWFZmI4XtLTd3VxMI4Drw1HqAdEbLyi+ctEY52HVu0CSnYGH6Bkn98nfYh1x3h IJqoOGXe3mg6umj+IbgfuiC0vw34UKPNV0kNQXvjWW8fGpuh5RZuGc7471CJkFhY 9ZNXXWpaaIanLl+CqkGRkjBVFJjZ2obsJa/jNlxAOzECELzAYSFKNUeNArzYMHJD YB/P3C6r8oXBElGcqBZSqZAEwTS6SFrHTF8Vzc98Uznj6dZSlwe10HKbrfGbJnOJ r2SWLUwJ8zrjoVFynSCIEZaLUWFsAeOGt88OSgToZJE/oVSuDxrxHbVpNCPiWHM= =hHw8 -END PGP SIGNATURE- -- _ -- 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] Problem with outbound dialing from remote phone
- wget won't work because the phone requires a user ID and password for access, but lynx or curl might do the trick. I was able to use lynx - couldn't figure out the voodoo for curl. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, October 14, 2011 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Un-top-posting and trimming cruft... On Fri, 14 Oct 2011, Adam Robins wrote: Thanks I will do that. The user is remote, so I must first RDP into her home network and do it from her PC. Since you have access to the Asterisk server's command line, you could use wget to retrieve the index page from the phone outputting to a file and scp the file back to you for your local viewing pleasure. -- 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
Re: [asterisk-users] Problem with outbound dialing from remote phone
I appreciate this real out-of-the-box thinking! I can just wait for her to go to dinner and get into her PC. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, October 14, 2011 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Un-top-posting and trimming cruft... On Fri, 14 Oct 2011, Adam Robins wrote: Thanks I will do that. The user is remote, so I must first RDP into her home network and do it from her PC. Since you have access to the Asterisk server's command line, you could use wget to retrieve the index page from the phone outputting to a file and scp the file back to you for your local viewing pleasure. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
Un-top-posting... On Fri, 14 Oct 2011, Steve Edwards wrote: Since you have access to the Asterisk server's command line, you could use wget to retrieve the index page from the phone outputting to a file and scp the file back to you for your local viewing pleasure. On Fri, 14 Oct 2011, Danny Nicholas wrote: - wget won't work because the phone requires a user ID and password for access, but lynx or curl might do the trick. I was able to use lynx - couldn't figure out the voodoo for curl. wget\ --output-document=index.html\ --password=pass\ --user=user\ http://a.b.c.d or wget http://user:pass@a.b.c.d -- 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
Re: [asterisk-users] Problem with outbound dialing from remote phone
On Fri, 14 Oct 2011, Danny Nicholas wrote: I was able to use lynx - couldn't figure out the voodoo for curl. curl\ --output index.html\ --user user:pass\ http://a.b.c.d -- 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
Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
On Thu, Oct 13, 2011 at 12:37:28AM +0200, Andreas Sikkema wrote: So normally calls to these destinations have nice caller id as if A was calling C (at least that's what C sees in their display) but every now and then I flow over to the alternative route and the information is lost, C doesn't see A, but B. Nothing I can do about it, been fighting over it for ages but I just doesn't seem to be able to make it work. Suddenly I feel very lucky. It only took me a couple of weeks of sending 10 test calls per day of resulting callerids mishaps with Verizon to get them to finaly trace the problem and correct a misconfigured switch. It also helps to be able to route all mobile traffic through an other provider, if they start to lose lots of minutes providers will act. -- Daniel Tryba -- _ -- 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] Problem with outbound dialing from remote phone
Did anything else change on her home network that could correlate to the time this started flaking on you? (eg: a new router/gateway) BB -- _ -- 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] Problem with outbound dialing from remote phone
I'm working that angle. I tried to use Dameware to get into her router via her home PC, but the screens weren't drawing correctly. I'll need to try LogmeIn. Also the IP address she read me directly off the phone is dubious. I cant ping it nor can I bring up the web interface. To be continued . . . -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Bosiljevac Sent: Friday, October 14, 2011 5:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Did anything else change on her home network that could correlate to the time this started flaking on you? (eg: a new router/gateway) BB -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Binding asterisk to two static IPs
On 10/12/11 2:27 PM, ge...@riseup.net wrote: If you put 0.0.0.0, it will bind to all addresses. In a HA Cluster, on the active node, if you have a box address of 192.168.1.101 and a floating address of 192.168.1.102, then if you use bindaddr=0.0.0.0 you will find that phones on the 192.168.1.x subnet will not register on the floating address, which of course defeats the point of HA clustering. What happens is that the registration packets go to the floating address 192.168.1.102 but the response packets appear to come from 192.168.1.101 [same NIC but the packet contains the base address attached to the NIC], so the registration fails. Any idea how to solve this? try use 2 different subnet addresses instead of both addresses on the same subnet. e.g. 192.168.1.101/24 192.168.2.101/24 and also use ip command to add the address to the interface instead of ifconfig and eth0:x notation. that way the OS will pick the correct address when responding to in coming packets. the problem is when you assign 2 addresses to the same interface on the same subnet, one of them will be primary and the other will becomes secondary on that subnet. the OS will always pick the primary address when sending out packets on that subnet. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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
