[asterisk-users] Transfer (sip -> dahdi) results in moh for dahdi
I have had this problem for a while, so I can't be sure when it started or what was changed. The setup is this: 2 sip handsets (a Cisco 7960 and a 7961) exten 401/402 1 fxs/dahdi cordless phone, exten 201 rhino fxo/fxs analog card asterisk 1.4.31 This is running on a Soekris 5501 with Astlinux 0.7.2 While I do have FXO capabilities, no POTS lines are connected. When a call comes in (VoIP, either SIP or IAX) it is usually answered on one of the SIP Cisco phones(x 401 or 402). If it is for my wife, which it usually is, and she is walking around the house, then I would like to transfer the call to the fxs/dahdi analog cordless phone (x 201). At one time this worked, but about a year or so ago it stopped. What is happening now is that the call comes in (x 401), is transferred via the cisco transfer soft button to (x 201), ... during this time the caller was put on hold or rather was automatically connected to the MOH process... , When (x 201) answers the phone, they are connected to the MOH process and cannot hear or talk to the original caller. In testing, if I leave the (x 201) call open, the original outside call is kept open as well. A look at the "active sessions" confirms this. When either (x 201) or original caller hang up, the call/connection is terminated. I can transfer calls from one Cisco to the other without issue; and if my laptop, with the softphone installed, had not just taken a turn for the worst, I would test Cisco to Bria/Counterpath and let you know how that would work. I have looked around at my configs, but don't see anything that would cause this... but truthfully I don't even know where to begin with something like this. I checked the logs to see if there was something helpful there but did not see anything. My only though is that it is something with the way the Cisco internal transfer process happens... but again, I don't know where to begin to test that theory. Has anyone seen or heard of this? Know how to resolve to expected behavior? I appreciate any pointers. John R. -- _ -- 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] pickup problem
Hi all, I can´t pickup calls on my asterisk. When I try to load app_pickupchan.so I receive following message: Module 'app_pickupchan.so' was not compiled with the same compile-time options as this version of Asterisk It was working fine until few time ago. What is going on? Thanks! 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
Re: [asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address?
On Sat, Dec 11, 2010 at 1:04 PM, Bruce B wrote: > Thanks for the confirmation. Do you have both LAN and WAN as outbound AON > like this: > WAN any * * * * * YES > LAN any * * * * * YES > ??? > I am stumped as to why pfSense behaves like this in this instance. > Thanks again. You only want one outbound NAT if you only have WAN and LAN interfaces. Mine is WAN 192.168.1.0/24 * * * * * YES Replace 192.168.1.0/24 with your internal network range. Ryan -- _ -- 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] Why does "sip show peers" show my router/gateway address as the client IP address?
Thanks for the confirmation. Do you have both LAN and WAN as outbound AON like this: WAN any * * * * * YES LAN any * * * * * YES ??? I am stumped as to why pfSense behaves like this in this instance. Thanks again. On Sat, Dec 11, 2010 at 12:34 PM, Ryan Wagoner wrote: > On Sat, Dec 11, 2010 at 11:45 AM, Bruce B wrote: > > Thanks for the feedback Ryan. > > Siproxd is not installed. I think Siproxd like you said just does the > > reverse meaning if phones are part of pfSense subnet then it connects to > > outside world. But in my case they are coming into Asterisk which is on > > pfSense subnet. I do have a static IP and it's set like: > > externip=34.34.34.34 > > localnet=192.168.5.0/255.255.255.0 > > Do you use pfSense for this same situation? Can you do a sip show peers > and > > let me know if you actually see the outside public IP addresses for the > > clients? Also how is your outbound NAT setup? AON? > > Thanks > > > > Yep I am using pfSense 1.2.3 with a static IP. I have port forwarded > UDP SIP and the UDP RTP port range to the private IP of the Asterisk > box. I have enabled manual outbound nat and configured the static port > option. If you use the automatic outbound nat it will randomize the > ports, which you don't want. My sip.conf looks like yours with the > externip and localnet set. When I do sip show peers I see the external > IP. > > Ryan > > -- > _ > -- 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] Why does "sip show peers" show my router/gateway address as the client IP address?
On Sat, Dec 11, 2010 at 11:45 AM, Bruce B wrote: > Thanks for the feedback Ryan. > Siproxd is not installed. I think Siproxd like you said just does the > reverse meaning if phones are part of pfSense subnet then it connects to > outside world. But in my case they are coming into Asterisk which is on > pfSense subnet. I do have a static IP and it's set like: > externip=34.34.34.34 > localnet=192.168.5.0/255.255.255.0 > Do you use pfSense for this same situation? Can you do a sip show peers and > let me know if you actually see the outside public IP addresses for the > clients? Also how is your outbound NAT setup? AON? > Thanks > Yep I am using pfSense 1.2.3 with a static IP. I have port forwarded UDP SIP and the UDP RTP port range to the private IP of the Asterisk box. I have enabled manual outbound nat and configured the static port option. If you use the automatic outbound nat it will randomize the ports, which you don't want. My sip.conf looks like yours with the externip and localnet set. When I do sip show peers I see the external IP. Ryan -- _ -- 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] Why does "sip show peers" show my router/gateway address as the client IP address?
Hi Again, Here is what I see which is wrong for Addr>IP and is fine for Reg. Contact parameter - In fact both parameters should show the public IP address: ** DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 192.168.0.1 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: SIP Options : (none) Codecs : 0xe (gsm|ulaw|alaw) Codec Order : (ulaw:20,alaw:20,gsm:20) Auto-Framing : No 100 on REG : No Status : OK (14 ms) Useragent: Linksys/WRP400-1.01.00 Reg. Contact : sip:5...@45.45.45.45:5060 Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs ** Regards, Bruce On Sat, Dec 11, 2010 at 10:15 AM, Ryan Wagoner wrote: > On Sat, Dec 11, 2010 at 3:06 AM, Bruce B wrote: > > Hi Everyone, > > I am using pfSense to do firewall and NAT on an Asterisk server. I have > > ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local > IP > > 192.168.5.5. However, when a user from outside using Linksys WRP400 ata > > connects to the Asterisk server and registers I see them as 192.168.1.1 > in > > the "sip show peers" command. In face, all many different of the Linksys > > WRP400 show the same. It seems that pfsense does something to the packets > > that when they reach Asterisk it thinks they are sent from the Gateway > > rather than the actual endpoint hence the calls are not reaching the > other > > side but registration is made. > > Any experience with this? > > Thanks > > Do you have the siproxd package installed on pfsense? It is suspossed > to handle registrations from multiple phones behind NAT. In your case > since the phones are external I would probably remove it if installed. > I haven't needed siproxd. > > Also on Asterisk set externip to your static IP in sip.conf. Or if you > don't have a static IP set externhost. You also need to configure > localnet. > > Ryan > > -- > _ > -- 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] Why does "sip show peers" show myrouter/gateway address as the client IP address?
Hi Wang, Did you mean to write a feedback? You sent an empty message. Regards, On Sat, Dec 11, 2010 at 11:56 AM, wrote: > > Sent from my “contract free” BlackBerry® smartphone on the WIND network. > > -Original Message- > From: Bruce B > Sender: asterisk-users-boun...@lists.digium.com > Date: Sat, 11 Dec 2010 11:45:15 > To: Asterisk Users Mailing List - Non-Commercial Discussion< > asterisk-users@lists.digium.com> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Why does "sip show peers" show my > router/gateway address as the client IP address? > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why does "sip show peers" show myrouter/gateway address as the client IP address?
Sent from my “contract free” BlackBerry® smartphone on the WIND network. -Original Message- From: Bruce B Sender: asterisk-users-boun...@lists.digium.com Date: Sat, 11 Dec 2010 11:45:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address? -- _ -- 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] Why does "sip show peers" show my router/gateway address as the client IP address?
Thanks for the feedback Ryan. Siproxd is not installed. I think Siproxd like you said just does the reverse meaning if phones are part of pfSense subnet then it connects to outside world. But in my case they are coming into Asterisk which is on pfSense subnet. I do have a static IP and it's set like: externip=34.34.34.34 localnet=192.168.5.0/255.255.255.0 Do you use pfSense for this same situation? Can you do a sip show peers and let me know if you actually see the outside public IP addresses for the clients? Also how is your outbound NAT setup? AON? Thanks On Sat, Dec 11, 2010 at 10:15 AM, Ryan Wagoner wrote: > On Sat, Dec 11, 2010 at 3:06 AM, Bruce B wrote: > > Hi Everyone, > > I am using pfSense to do firewall and NAT on an Asterisk server. I have > > ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local > IP > > 192.168.5.5. However, when a user from outside using Linksys WRP400 ata > > connects to the Asterisk server and registers I see them as 192.168.1.1 > in > > the "sip show peers" command. In face, all many different of the Linksys > > WRP400 show the same. It seems that pfsense does something to the packets > > that when they reach Asterisk it thinks they are sent from the Gateway > > rather than the actual endpoint hence the calls are not reaching the > other > > side but registration is made. > > Any experience with this? > > Thanks > > Do you have the siproxd package installed on pfsense? It is suspossed > to handle registrations from multiple phones behind NAT. In your case > since the phones are external I would probably remove it if installed. > I haven't needed siproxd. > > Also on Asterisk set externip to your static IP in sip.conf. Or if you > don't have a static IP set externhost. You also need to configure > localnet. > > Ryan > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No more room in scheduler
On Sat, Dec 11, 2010 at 6:40 AM, mahfoudh alfaqeeh wrote: > Dears: > > Really, later I faced problem in the asterisk system which is : > Message is shown when the unique id which is generated with each caller > reach > 9000 and something: > > No more room in scheduler > Asked to delete sched id > . > . > > after I restarted the server this message is not shown again till now > (after 2 week) > >>> > My question: > What is the reason of this error and how can I solve the problem > permanently > > Please I need the help as soon as possible. > Your effort is appreciated>>> > Thank So much.. > Tell us your exact version asterisk. Cut and paste the actual, exact messages you see. "No more room in scheduler" doesn't show up in 1.6.2, or trunk. Schedulers don't have any limits; the size of the uniqueid's don't either. Try "core show threads", "core show calls" "sip show channels" "core show taskprocessors" from within asterisk's cli. Try this outside asterisk: "lsof -p | wc" (you may have to install lsof via your package manager) > > > > > > > -- > _ > -- 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 > Steve Murphy ParseTree Corp. 57 Lane 17 Cody, WY 82414 ✉ m...@parsetree.com ☎ 307-899-5535 -- _ -- 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] Why does "sip show peers" show my router/gateway address as the client IP address?
On Sat, Dec 11, 2010 at 3:06 AM, Bruce B wrote: > Hi Everyone, > I am using pfSense to do firewall and NAT on an Asterisk server. I have > ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local IP > 192.168.5.5. However, when a user from outside using Linksys WRP400 ata > connects to the Asterisk server and registers I see them as 192.168.1.1 in > the "sip show peers" command. In face, all many different of the Linksys > WRP400 show the same. It seems that pfsense does something to the packets > that when they reach Asterisk it thinks they are sent from the Gateway > rather than the actual endpoint hence the calls are not reaching the other > side but registration is made. > Any experience with this? > Thanks Do you have the siproxd package installed on pfsense? It is suspossed to handle registrations from multiple phones behind NAT. In your case since the phones are external I would probably remove it if installed. I haven't needed siproxd. Also on Asterisk set externip to your static IP in sip.conf. Or if you don't have a static IP set externhost. You also need to configure localnet. Ryan -- _ -- 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 more room in scheduler
Dears: Really, later I faced problem in the asterisk system which is : Message is shown when the unique id which is generated with each caller reach 9000 and something: No more room in scheduler Asked to delete sched id . . after I restarted the server this message is not shown again till now (after 2 week) >>> My question: What is the reason of this error and how can I solve the problem permanently Please I need the help as soon as possible. Your effort is appreciated>>> Thank So much.. -- _ -- 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] Why does "sip show peers" show my router/gateway address as the client IP address?
Hi Everyone, I am using pfSense to do firewall and NAT on an Asterisk server. I have ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local IP 192.168.5.5. However, when a user from outside using Linksys WRP400 ata connects to the Asterisk server and registers I see them as 192.168.1.1 in the "sip show peers" command. In face, all many different of the Linksys WRP400 show the same. It seems that pfsense does something to the packets that when they reach Asterisk it thinks they are sent from the Gateway rather than the actual endpoint hence the calls are not reaching the other side but registration is made. Any experience with this? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
