[Asterisk-Users] AGI or somthing else??
Hi, I know that AGI can be used to determine variables from external sources, but AGI can only be used from the extensions.conf file. Is there anyway to retrieve external information for the sip.conf file?? Here is why.. I will have users on remote networks with dynamic IP addys.. I have an application that can log the IP address of the person that sucessfully logs into it.. So I want to tie the app system together with the PBX in that by authenticating to the app a user will be granted access to the PBX using a SIP phone based on the IP address, so users don't have to remember another password. Any ideas how this could be done? Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Whoah! My E400P system went AWOL
Hi, I came back from a quiet weekend today and found my E400P box to have gone astray. Asterisk is loaded from inittab, and started crashing and reloading a couple of thousands of times, each time notifying my monitoring service :-P I remember there would be issues on old cvs stuff since the crash at digium so I made a clean checkout just now. Here is what happens when I load manually: [chan_oh323.so]WARNING[11276]: File chan_zap.c, Line 5237 (zt_pri_error): PRI: Read on 140 failed: Unknown error 500 PRI got event: 5 NOTICE[12301]: File chan_zap.c, Line 4233 (handle_init_event): Alarm cleared on channel 94 (last line repeated for every defined channel) == D-Channel on span 2 up == Restart on requested on entire span 2 /usr/local/sbin/astrun: line 4: 329 Segmentation fault /usr/sbin/asterisk -c Ouch ... error while writing audio data: : Broken pipe vectra:~# Any clues ? zttool sees the channel status OK. Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI or somthing else??
The situation I was trying to get to was that the Asterisk SIP authentication would be based soley on IP address and that my other app would provide the user validation.. That way I could map an extension to a user and it would not matter which PC they logged into their calls would be routed to the right place.. In thinking about it, it would not work anyway seeing as the conf files are all read into the system on startup so a change in the sip.conf would not take effect until the system was reloaded.. Which brings me back to the original problem of allowing muliple users on the same workstation without having to teach them how to reconfigure the IP phone to recieve calls on their own extension?? - Original Message - From: Steven Critchfield [EMAIL PROTECTED] Date: Sun, 23 Mar 2003 10:09:36 -0600 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AGI or somthing else?? On Sun, 2003-03-23 at 08:03, WipeOut . wrote: Hi, I know that AGI can be used to determine variables from external sources, but AGI can only be used from the extensions.conf file. Is there anyway to retrieve external information for the sip.conf file?? Here is why.. I will have users on remote networks with dynamic IP addys.. I have an application that can log the IP address of the person that sucessfully logs into it.. So I want to tie the app system together with the PBX in that by authenticating to the app a user will be granted access to the PBX using a SIP phone based on the IP address, so users don't have to remember another password. 2 options here. Easiest: The sip software/device should only need to be configured once no matter what IP address it is on and will login itself. So no need to remember new password. Harder but not really that hard: Lock down your asterisk ports with linux firewall capabilities and when your app logs the user in, it applies a rule allowing that user into the aterisk machine. Of course if you plan to route calls to the user, they need to have at least a username configured to the sip device so it can identify the user. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI or somthing else??
On Sun, 2003-03-23 at 12:38, WipeOut . wrote: The situation I was trying to get to was that the Asterisk SIP authentication would be based soley on IP address and that my other app would provide the user validation.. That way I could map an extension to a user and it would not matter which PC they logged into their calls would be routed to the right place.. In thinking about it, it would not work anyway seeing as the conf files are all read into the system on startup so a change in the sip.conf would not take effect until the system was reloaded.. Which brings me back to the original problem of allowing muliple users on the same workstation without having to teach them how to reconfigure the IP phone to recieve calls on their own extension?? Are these hardware SIP phones, or are they software phones? If software, you should be able to have unique configurations based upon unique logins to the PC. If hardware, you can assign specific username/password combos to the phone, and do lookups via AGI to a file that will link user to phone. For perl, you can use the freezthaw module to store the data fast and retrieve it fast. You could have a hash of your users, and assign the phones username to the user. Then the AGI app could dial this SIP phone upon lookup. The benefit here is you can have a perl app external of asterisk to also manipulate the file such as login/assign sip phone, logout/unassign sip phone. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAL - Sip Extentions
Hi All, Im very new to Asterisk, so can someone give me the context to use In the Extentions.conf file to ring a SIP Phone? (Cisco 7960) Also, does each IP Phone (again, Cisco 7960) need a separate entry In the SIP.CONF File? Thanks a ton! Ben Bawkon P.S. a sample SIP.CONF file that works well for multiple DHCPd Cisco 7960s Would be really nice to be able to look over.
Re: [Asterisk-Users] SIP and rfc2833 dtmf + # transfer
Yes. Lele Forzani wrote: Has anybody noticed that # transfers aren't working anymore when SIP is used with rfc2833 dtmfmode? They work as espected with inband dtmf. lele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Luke, here's some information I got back from iconnect: 1) the prefix is not a toggle. It tells iconnects SIP gateway to use compressed codecs. The choices are gsm, g723.1, g729. If you don't use , the gateway will tried to use PCMu/8000 (ulaw?) or PCMa/8000 (alaw?). I can get the gateway to work with g723.1 and gsm, but I can't get it to work with ulaw or alaw. My phone device is a quicknet linecard. The g723.1 format on the linecard does not work with iconnect. If I use it then the audio to and from iconnect is distorted (as if it is using the wrong format or has sampling errors). Gregg On Thu, 2003-03-20 at 18:25, Luke Howard wrote: I've found the same. If I make an outgoing call (snom 200 handset), I get about 5 seconds of audio and then it drops out (very occasionally it does work). Incoming calls appear to work, though. -- Executing Goto(SIP/515-Office-143b, iconnecthere-ulaw|91800XXX|1) in new stack -- Goto (iconnecthere-ulaw,91800XXX,1) -- Executing StripMSD(SIP/515-Office-143b, 1) in new stack -- Executing Dial(SIP/515-Office-143b, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnecthere-960b answered SIP/515-Office-143b -- Attempting native bridge of SIP/515-Office-143b and SIP/iconnecthere-960b -- Got SIP response 480 Temporarily not available back from 213.137.73.178 == Spawn extension (iconnecthere-ulaw, 1800XXX, 2) exited non-zero on 'SIP/515-Office-143b' SIP config is: [general] port=5060 bindaddr=0.0.0.0 context=sip-remote disallow=all allow=ulaw allow=alaw tos=lowdelay tos=184 register = 1XX:[EMAIL PROTECTED] [iconnecthere] type=friend username= password= host=sipauth.deltathree.com context=iconnecthere-ulaw callerid=PADL Software Pty Ltd (XXX) XXX ;txgain = 5.0; ;rxgain = 5.0; inbanddtmf=1 -- Luke P.S. Is anyone planning on licensing G 723.1 for use with Asterisk? As I understand it, buying a LineJACK won't suffice if the card's DSP is not actually used. -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast_rtp_write
mark, How come there is no entry in the case statement in ast_rtp_write to handle G723? If I get G723 to work, asterisk starts spitting out lots of warnings about Don't know how to send format %d packets with RTP. Gregg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Convert you FXS port to FXO cheap
If you have an FXS port and would like to attach a PSTN analog line to it this device would do the job by converting the FXS port to FXO. Its a small external device. Works well with VOIP FXS and other FXS interfaces. Interface: 2 RJ11 Jacks (one for the FXS port and one for the PSTN outlet.) Cost:$35.00 with USPS regular mail included. Power: 9 to 20V dc power supply (Not included) Thanks Dave
[Asterisk-Users] Convert yourr FXS port to FXO cheap
If you have an FXS port and would like to attach a PSTN analog line to it this device would do the job by converting the FXS port to FXO. Its a small external device. Works well with VOIP FXS interfaces and other FXS interfaces. Interface: 2 RJ11 Jacks (one for the FXS port and one for the PSTN outlet.) Cost:$35.00 with USPS regular mail included. Power: 9 to 20V dc power supply (Not included) Thanks Dave
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Gregg, 1) the prefix is not a toggle. It tells iconnects SIP gateway to use compressed codecs. The choices are gsm, g723.1, g729. I figured as much. I'm sticking with G.711 as GSM sounds horrible (at least with the snom phones) and the other codecs you mention are patent encumbered. I can get the gateway to work with g723.1 and gsm, but I can't get it to work with ulaw or alaw. My phone device is a quicknet linecard. The problem I'm having appears to be purely a signalling one. -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Standalone S100U: Is there some trick?
Not sure how I can look inside to see what asterisk could be doing. . While it's running, you can do ps auxww | grep asterisk. Look for the PID of the process taking the CPU. For example: root 1206 0.0 0.9 118576 4904 pts/2 SMar21 0:00 ./asterisk -vvvgc . . . root 1216 0.0 0.9 118576 4904 pts/2 SMar21 1:52 ./asterisk -vvvgc This thread has taken 1:52 in CPU time. So, we now run gdb on the *first* PID of asterisk, *not* necessarily the with the highest CPU... # gdb ./asterisk 1206 And you'll eventulaly be dumped at something like this: 0x40188544 in __libc_read () from /lib/i686/libc.so.6 (gdb) Now type info threads to see all the threads. The one with the matching LWP is the one you want. For example: 9 Thread 7176 (LWP 1216) 0x4018d9f7 in __poll (fds=0x80d5ef0, nfds=1, timeout=1000) at ../sysdeps/unix/sysv/linux/poll.c:63 Now switch like this: (gdb) thread 9 and now bt for backtrace (gdb) bt And you'll get something like this: #0 0x4018d9f7 in __poll (fds=0x80d5ef0, nfds=1, timeout=1000) at ../sysdeps/unix/sysv/linux/poll.c:63 #1 0x08050adc in ast_io_wait (ioc=0x80d4e00, howlong=1000) at io.c:255 #2 0x4330b96d in network_thread (ignore=0x0) at chan_iax.c:4274 #3 0x40035b9c in pthread_start_thread (arg=0x43c11be0) at manager.c:274 (gdb) That tells us the thread in question is the IAX network thread runing in chan_iax. Does that help? Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Convert you FXS port to FXO cheap
Do you have a picture available somewhere on the net ? Dave George wrote: MessageRay, The name of the device is FXS to FXO converter. You need a device for each port. Thanks Dave - Original Message - From: Ray Dzek To: [EMAIL PROTECTED] Sent: Sunday, March 23, 2003 2:47 PM Subject: Re: [Asterisk-Users] Convert you FXS port to FXO cheap Well the obvious question is... Is this something that is supported in *? So for instance, if I have an 8 port MGCP VOIP box, can I convert a couple of those to FXO and have * utilize them? - Original Message - From: Dave George To: [EMAIL PROTECTED] Sent: Sunday, March 23, 2003 2:04 PM Subject: Re: [Asterisk-Users] Convert you FXS port to FXO cheap Hi, I am the retailer. You can also purchase whole sale from me. This device is manufactured by my company. Thanks Dave - Original Message - From: Abdul Hakeem To: [EMAIL PROTECTED] Sent: Sunday, March 23, 2003 1:28 PM Subject: RE: [Asterisk-Users] Convert you FXS port to FXO cheap Hi, Could you pass on the details of the retailer/manufacturer ? Cheers, Abdul Hakeem -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, March 23, 2003 8:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Convert you FXS port to FXO cheap If you have an FXS port and would like to attach a PSTN analog line to it this device would do the job by converting the FXS port to FXO. It's a small external device. Works well with VOIP FXS and other FXS interfaces. Interface: 2 RJ11 Jacks (one for the FXS port and one for the PSTN outlet.) Cost: $35.00 with USPS regular mail included. Power: 9 to 20V dc power supply (Not included) Thanks Dave -- Thanks, Jean-Pierre Denis jp at msfree dot ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a problem with MeetMe
That is to say,if I have no zaptel device installed in my * box,then I could'nt use the meetme function? john - Original Message - From: Brancaleoni Matteo [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, March 04, 2003 9:32 AM Subject: Re: [Asterisk-Users] a problem with MeetMe have you any zaptel device in your box? a zaptel device is required for timing source for the conference (so meetme) matteo Il mar, 2003-03-04 alle 16:49, Rattana BIV ha scritto: Hi, I try the application MeeMe but i Have a problem when I call a conference. It show me : Unable to open pseudo channel Does anyone can help me ? regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P incoming call handling
quick quess is you are not answering the line. exten = s,1,Answer exten = s,2,Dial,SIP/snomphone On Sun, 2003-03-23 at 17:05, Tamas Levente wrote: Hi, I have some problem with a x100p hadling incoming calls. It goes to the incoming context, which looks like: exten = s,1,Dial,SIP/snomphone I have [snomphone] in sip.conf. So when call comes in, it rings on the snomphone, but I can't answer the call. I get the voip phone and after a few noises it says released. But the line keep ringing in Zap. Please help __ Levente Tamás ICQ#: 13692773 Current ICQ status: [image] + More ways to contact me __ -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] will the digium cards
Digium has provided some cards to the David Sugar of the Bayonne project, as well as made ourselves available for support, but as far as I know, they have not added support for the boards yet. I would suggest contacting David Sugar for more information. Mark On Sat, 22 Mar 2003, d hinton wrote: work with Bayonne? will i need special drivers to use them with Bayonne? thanks 4 your help dwayne ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linux Kernel Patch
Does anyone know the location of the kernel patch to disable isdn dtmf detection? Also the location of the asterisk patch for doing the dtmf detection? Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Is there a Record-Route header in the response that comes back from iconnect? Mark On Sun, 23 Mar 2003, Luke Howard wrote: Or maybe we should send an ACK to them -- I need to read the SIP RFC... Tried that, doesn't work. I should add that in my config I'm totally behind NAT, both asterisk and an ATA186 that talks to it. So that may be confounding me in terms of what I'm seeing. I do see the same problem: after a few minutes, the call is dropped (this is using Asterisk patched to ignore 480 Temporarily not available errors). From the log below it _seems_ like iConnectHere is waiting for an acknowledgment to the 480, but you noted that you tried this? It seems to be purely a signalling problem as the call is setup fine between the SIP phone and the gateway (which in this case appeared to be somewhere in Austria...) -- Executing Macro(SIP/515-Office-b8b1, iconnecthere|33145207135|60) in new stack -- Executing Dial(SIP/515-Office-b8b1, SIP/[EMAIL PROTECTED]|60|r) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnecthere-ec91 is ringing -- SIP/iconnecthere-ec91 answered SIP/515-Office-b8b1 -- Attempting native bridge of SIP/515-Office-b8b1 and SIP/iconnecthere-ec91 -- Got SIP response 408 Request Timeout back from 213.137.73.178 == Spawn extension (macro-iconnecthere, s, 1) exited non-zero on 'SIP/515-Office-b8b1' in macro 'iconnecthere' == Spawn extension (local, 933145207135, 1) exited non-zero on 'SIP/515-Office-b8b1' -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Mark, I believe there is: Here is the exchange using sip debug. Gregg --- bigcat*CLI sip debug SIP Debugging Enabled -- Executing Dial(Phone/phone0, SIP/[EMAIL PROTECTED]) in new stack Interface is eth0 IP Address is 192.168.4.3 We're at 192.168.4.3 port 39998 Answering with preferred capability 4 Answering with preferred capability 8 Answering with preferred capability 1 Answering with preferred capability 2 10 headers, 10 lines XXX Need to handle Retransmitting XXX: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e Contact: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 202 v=0 o=root 13858 13858 IN IP4 192.168.4.3 s=session c=IN IP4 192.168.4.3 t=0 0 m=audio 39998 RTP/AVP 0 8 4 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 (no NAT) to 213.137.73.178:5060 -- Called [EMAIL PROTECTED] Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 Call-ID: [EMAIL PROTECTED] From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 7 headers, 0 lines Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 Call-ID: [EMAIL PROTECTED] From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED];tag=4b857f0d-5ab788dc CSeq: 102 INVITE Proxy-Authenticate: DIGEST realm=deltathree.com, nonce=3e7e6ed6, algorithm=MD5 Content-Length: 0 8 headers, 0 lines XXX Need to handle Retransmitting XXX: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED];tag=4b857f0d-5ab788dc Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 213.137.73.178:5060 We're at 192.168.4.3 port 39998 Answering with preferred capability 4 Answering with preferred capability 8 Answering with preferred capability 1 Answering with preferred capability 2 XXX Need to handle Retransmitting XXX: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e Contact: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username=85904362, realm=deltathree.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=3e7e6ed6, response=b6bab0a7e409d10496cd6140e6d1e063 Content-Type: application/sdp Content-Length: 202 v=0 o=root 13837 13837 IN IP4 192.168.4.3 s=session c=IN IP4 192.168.4.3 t=0 0 m=audio 39998 RTP/AVP 0 8 4 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 (no NAT) to 213.137.73.178:5060 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 Call-ID: [EMAIL PROTECTED] From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED] CSeq: 103 INVITE Content-Length: 0 7 headers, 0 lines Sip read: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0;received=66.30.28.60 From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED];tag=1FD16250-8D Date: Mon, 24 Mar 2003 02:35:05 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 180 v=0 o=CiscoSystemsSIP-GW-UserAgent 3626 3613 IN IP4 213.137.65.239 s=SIP Call c=IN IP4 213.137.65.239 t=0 0 m=audio 18358 RTP/AVP 4 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no 12 headers, 8 lines Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0;received=66.30.28.60 From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED];tag=1FD16250-8D Date: Mon, 24 Mar 2003 02:35:05 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: sip:[EMAIL PROTECTED]:5060 Record-Route: sip:213.137.79.80, sip:213.137.79.78, sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176 Content-Type: application/sdp Content-Length: 180 v=0 o=CiscoSystemsSIP-GW-UserAgent 3626 3613 IN IP4 213.137.65.239 s=SIP Call c=IN IP4 213.137.65.239 t=0 0 m=audio 18358 RTP/AVP 4 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no 14 headers, 8 lines XXX Need to handle Retransmitting XXX: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED];tag=4b857f0d-5ab788dc Call-ID: [EMAIL PROTECTED] CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 213.137.73.178:5060 --
Re: [Asterisk-Users] X100P incoming call handling
It *shouldn't* be necessary to answer the line before executing dial. At the point the SIP phone answers, it should *then* answer the X100P. As usual, find me on IRC and I can spend some time debugging it. See http://www.digium.com for IRC info. Mark On Sun, 23 Mar 2003, Steven Critchfield wrote: quick quess is you are not answering the line. exten = s,1,Answer exten = s,2,Dial,SIP/snomphone On Sun, 2003-03-23 at 17:05, Tamas Levente wrote: Hi, I have some problem with a x100p hadling incoming calls. It goes to the incoming context, which looks like: exten = s,1,Dial,SIP/snomphone I have [snomphone] in sip.conf. So when call comes in, it rings on the snomphone, but I can't answer the call. I get the voip phone and after a few noises it says released. But the line keep ringing in Zap. Please help __ Levente Tamás ICQ#: 13692773 Current ICQ status: [image] + More ways to contact me __ -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAL - Sip Extentions
Hi Ben, Try Dial(SIP/ben) Make sure you have a sip.conf entry for ben, or whatever you want. Yes, each phone needs an entry. If you need a example, let me know. --Mike On Sunday, March 23, 2003, at 11:46 AM, Benjamin J. Bawkon wrote: Hi All, Im very new to Asterisk, so can someone give me the context to use In the Extentions.conf file to ring a SIP Phone?(Cisco 7960) ? Also, does each IP Phone (again, Cisco 7960) need a separate entry In the SIP.CONF File? ? Thanks a ton! Ben Bawkon ? P.S. a sample SIP.CONF file that works well for multiple DHCPd Cisco 7960s Would be really nice to be able to look over.
Re: [Asterisk-Users] Convert you FXS port to FXO cheap
On Mon, 2003-03-24 at 00:19, Ray Dzek wrote: Umm ... nothing personal, but that still didn't answer the question of if it will work, or is it supported by *. There is no reason why it shouldn't work with asterisk. The point of the device is to convert a FXS port to a FXO port. Which means it probably detects the ring on a FXO port, then loops the FXS port to signal the line as picked up, then connects the 2 lines together. So all asterisk needs to do is detect the loop and instead of providing dialtone, it needs to act like it just answered a call coming in like a hotline phone. See archives about the hotline phone last week. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie please help!
On Sun, 23 Mar 2003, Gary Ruddock wrote: I wanted to get the E100 digium hardware and use asterix but what concerns me is the telephonists interface. What software is available that links with asterix, can display a list of wating calls and will allow me to open a dynamically generated URL in a browser. i presume you mean the client interface, at the moment you have a number of options, but the best one is gnophone, since it has mozilla-embedded and you can open any webpage, and pass information like CLID to it, moreover its designed for * :) you could also use any other SIP / IAX / H.323 client and hack that together with a http or other browser however, the E100 interface will only come into play if you have an E1 coming in from the telco... if you have POTS lines (standard analog phone lines) then for four lines you might want to put 4 x X100P PCI cards, or invest in a T100P and a channel bank (since the T1 channel banks are far cheaper) I gave seen gnophone and gnomeeting. Could I get asterix to send a url and telephone to a voip client like gnophone and could the gnophone user then click that link and answer at the same time maybe? in a word, yes, * is a pretty good option for a call-center environment, afaik, the only decent open-source option -- Mirza Wasim Baig | Principal Consultant | Convergence Business Systems VOX: +92(51)282-0628 x7400 | FAX: +92(51)282-0621 | IAX: (700)282-0628 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Convert you FXS port to FXO cheap
would be nice is their was a picutre and some details somewhere ? On Mon, 24 Mar 2003 00:55:33 -0600, Steven Critchfield wrote: On Mon, 2003-03-24 at 00:19, Ray Dzek wrote: Umm ... nothing personal, but that still didn't answer the question of if it will work, or is it supported by *. There is no reason why it shouldn't work with asterisk. The point of the device is to convert a FXS port to a FXO port. Which means it probably detects the ring on a FXO port, then loops the FXS port to signal the line as picked up, then connects the 2 lines together. So all asterisk needs to do is detect the loop and instead of providing dialtone, it needs to act like it just answered a call coming in like a hotline phone. See archives about the hotline phone last week. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Convert you FXS port to FXO cheap
Okay then.. the next silly question is: Will using the FXS-FXO Converter off my VOIP gateway fix the echo I have now from running X100P for my PSTN connection? Thanks! Ray - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, March 23, 2003 10:55 PM Subject: Re: [Asterisk-Users] Convert you FXS port to FXO cheap On Mon, 2003-03-24 at 00:19, Ray Dzek wrote: Umm ... nothing personal, but that still didn't answer the question of if it will work, or is it supported by *. There is no reason why it shouldn't work with asterisk. The point of the device is to convert a FXS port to a FXO port. Which means it probably detects the ring on a FXO port, then loops the FXS port to signal the line as picked up, then connects the 2 lines together. So all asterisk needs to do is detect the loop and instead of providing dialtone, it needs to act like it just answered a call coming in like a hotline phone. See archives about the hotline phone last week. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
