Re: [asterisk-users] Incoming calls get 488 error
Hi, By the sip trace is very difficult to tell because the SIP messages are fine. Try to enable all codec, and if possible copy and paste your asterisk sip configuration for this peer. Enviado do meu telefone Android usando o Symantec TouchDown (www.symantec.com) -Original Message- From: Technical Support [supp...@telium.ca] Received: sexta-feira, 21 ago 2015, 19:46 To: asterisk-users@lists.digium.com [asterisk-users@lists.digium.com] Subject: [asterisk-users] Incoming calls get 488 error I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset. I captured the SIP traffic and see that my M65 is replying with an 488 not acceptable here. From what I read this is usually codec related but both asterisk and the M65 are set for ulaw as first choice. I have a SIP trace below. Can someone suggest why the 488 is being generated? --- Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (1198 bytes) INVITE sip:290006@192.168.253.20;line=14994 SIP/2.0 Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a Max-Forwards: 70 From: test user sip:230@192.168.253.4;tag=as7b616c8d To: sip:290006@192.168.253.20;line=14994 Contact: sip:230@192.168.253.4:5060 Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060 CSeq: 102 INVITE User-Agent: FPBX-2.11.0(11.10.2) Date: Fri, 21 Aug 2015 22:37:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 606 v=0 o=root 1678280845 1678280845 IN IP4 192.168.253.4 s=Asterisk PBX 11.10.2 c=IN IP4 192.168.253.4 b=CT:384 t=0 0 m=audio 18090 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 12226 RTP/AVP 99 98 34 31 a=rtpmap:99 H264/9 a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0 a=rtpmap:98 H263-1998/9 a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 a=rtpmap:34 H263/9 a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 a=rtpmap:31 H261/9 a=sendrecv Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (280 bytes) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a From: test user sip:230@192.168.253.4;tag=as7b616c8d To: sip:290006@192.168.253.20;line=14994 Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060 CSeq: 102 INVITE Content-Length: 0 Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (441 bytes) SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a Max-Forwards: 70 From: test user sip:230@192.168.253.4;tag=as7b616c8d To: sip:290006@192.168.253.20;line=14994;tag=ld65q Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060 CSeq: 102 INVITE Contact: sip:290006@192.168.253.20;line=14994 User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 0; HW=255) Content-Length: 0 -- _ -- 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 running on a Xen Centos Serverchallenge!!!
There's something weird at you linux system. Your kernel sources are i686 sources, but the output of your ls shows you are on a X86-64 system. (?) lrwxrwxrwx 1 root root 54 Nov 6 23:31 build - ../../../usr/src/kernels/2.6.18-164.6.1.el5-xen-x86_64 Look here! -bash-3.2# ls /usr/src/kernels 2.6.18-164.6.1.el5-xen-i686 2.6.18-164.6.1.el5xen-i686 and here! To install (and update) the correct kernel and kernel sources, try using: yum install kernel-* After that, reboot with the new kernel and recompile Dahdi. Rafael Prado +55 (11) 3323-1055 ttp://www.practis.com.br PRACTIS - Comunicação Tecnologia Av Aquidaban, 766 - Conj 51 CEP 13026-510, Campinas/SP - Brasil -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Daniel Leite de Abreu Sent: domingo, 21 de março de 2010 10:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk running on a Xen Centos Serverchallenge!!! Hi Thanks very much for reply it and helping me out. This is the out put -bash-3.2# ls -l /lib/modules/2.6.18-164.6.1.el5xen/ total 1280 lrwxrwxrwx 1 root root 54 Nov 6 23:31 build - ../../../usr/src/kernels/2.6.18-164.6.1.el5-xen-x86_64 drwxr-xr-x 2 root root 4096 Nov 3 17:31 extra drwxr-xr-x 9 root root 4096 Nov 6 23:31 kernel -rw-r--r-- 1 root root 282675 Nov 6 23:31 modules.alias -rw-r--r-- 1 root root 69 Nov 6 23:31 modules.ccwmap -rw-r--r-- 1 root root 231095 Nov 6 23:31 modules.dep -rw-r--r-- 1 root root147 Nov 6 23:31 modules.ieee1394map -rw-r--r-- 1 root root375 Nov 6 23:31 modules.inputmap -rw-r--r-- 1 root root 12632 Nov 6 23:31 modules.isapnpmap -rw-r--r-- 1 root root 74 Nov 6 23:31 modules.ofmap -rw-r--r-- 1 root root 219500 Nov 6 23:31 modules.pcimap -rw-r--r-- 1 root root 4033 Nov 6 23:31 modules.seriomap -rw-r--r-- 1 root root 132264 Nov 6 23:31 modules.symbols -rw-r--r-- 1 root root 356940 Nov 6 23:31 modules.usbmap lrwxrwxrwx 1 root root 5 Nov 6 23:31 source - build drwxr-xr-x 2 root root 4096 Nov 3 17:31 updates drwxr-xr-x 2 root root 4096 Nov 3 17:31 weak-updates -bash-3.2# This is the other out put. -bash-3.2# ls /usr/src/kernels 2.6.18-164.6.1.el5-xen-i686 2.6.18-164.6.1.el5xen-i686 waiting for you . Thanks very much daniel On 19 Mar 2010, at 8:51 PM, Tzafrir Cohen wrote: On Fri, Mar 19, 2010 at 01:26:43AM +0200, Tzafrir Cohen wrote: On Thu, Mar 18, 2010 at 05:03:12PM -0500, Warren Selby wrote: On Thu, Mar 18, 2010 at 6:56 PM, Daniel Leite de Abreu dlab...@gmail.comwrote: Hi David! Thanks very much for helping me out will all ! Ok i try your tip and @ the moment i still have the same problem but now i have the kernel and the kernel devel the same but wend i try to run make i still get the same erro, do you guys have any idea how to fix it? -bash-3.2# rpm -qa kernel* kernel-xen-devel-2.6.18-164.6.1.el5 kernel-xen-2.6.18-164.6.1.el5 -bash-3.2# After you install the kernel source, you'll need to rerun ./configure. Nope. The dahdi-linux makefile has no ./configure . You may want to run make clean and / or make distclean before rerunning ./configure. Specifically: it will look for: /lib/modules/VERSION/build/.config Where: VERSION is the kernel version string. 2.6.18-164.6.1.el5 in your case. 'build' is a symbolic link to the (often partial) kernel tree. What is the output of: ls -l /lib/modules/2.6.18-164.6.1.el5 What is the output of: ls /usr/src/kernels -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Using asterisk as avaya definity recordingserver
Hi, it's not that simple.
It requires deep modification on asterisk and dahdi sources to work the way
you want.
Rafael Prado
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Muro, Sam
Sent: quarta-feira, 17 de março de 2010 1:42
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Using asterisk as avaya definity
recordingserver
Hi there
Looks like someone hasnt done this!! I have been thinking and find out
that Monitor/Spy and the likes wont help me as the call need to be
bridged with the asterisk core or via channel drivers.
My final shot now is on Record() function. Since the legacy system will
forward the call to the monitoring interfaces when bridged within
itself, it the interface in on Asterisk, then we can capture the
pattern and use
exten =
#CALLER_NUMBER#CALLED_NUMBER,1,Record(/var/spool/asterisk/monitor/avaya
-${EXTEN:1:4}-${EXTEN:4:4}:wav)
This assume that Len(CALLER_NUMBER) = 4
Anyone with alternative solution?
Muro, Sam wrote:
Oh.. I didnt know that.
Thanks
Sam
Muro, Sam escribió:
What do you mean chief? What am looking at is ability for asterisk
to
receive a call and recording until it tier down without bridging it
to the physical device
Sam
Would you like the advice in all caps?
He means that you put the subject in all caps. He normally gets
upset
with everyone that does this on the subject or in the body. I've
corrected the caps in the subject to avoid further upsetting.
Cheers,
--
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Re: [asterisk-users] Transcoding question
How many simultaneous channels? Rafael Prado -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: segunda-feira, 22 de março de 2010 2:33 To: Asterisk User MailList Subject: [asterisk-users] Transcoding question We are getting ready to install a client that uses g729 when talking to their SIP provider to minimize bandwidth usage. We are going to want to be able to record the calls using AMI monitor actions into wav sound files. All the phones are soft phone running on Windows XP systems. Questions I have are what would the best codec be to have the soft phone use since, as I understand it, in order to mix the audio something will need to be transcoded. Can a two CPU quad core xeon 2GHz system handle the transcoding load or would if be better to have a daughter card handle the transcoding. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic 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] PGSQL application
Or you can write your own application module. Try looking at cdr_pgsql sources. ;) Rafael Prado +55 (11) 3323-1055 www.practis.com.br PRACTIS - Comunicação Tecnologia Av Aquidaban, 766 - Conj 51 CEP 13026-510, Campinas/SP - Brasil -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Vinícius Fontes Sent: quarta-feira, 10 de março de 2010 8:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PGSQL application - Tilghman Lesher tles...@digium.com escreveu: On Wednesday 10 March 2010 14:32:56 Vinícius Fontes wrote: Does the application PGSQL has been removed from Asterisk? Couldn't find it on Asterisk source and addons. That application has never been a part of Asterisk in the first place. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org Hmm I could swear I used it on the 1.2 days. So, in order to access PostgreSQL directly from the dialplan without the use of AGIs, much like the MYSQL() app, the only way to go is via the function ODBC()? -- _ -- 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 smime.p7s Description: S/MIME cryptographic 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
[asterisk-users] RES: Dahdi Congestion status
Yes, it will catch congestion cases on the receiver side. There's two ways to avoid it: 1) use ChanIsAvail on the dadhi channel. It works very well. Or 2) Create a special signal for dahdi congestion (require modifying the source and recompile) that differs from the congestion cause. Here we created code 130, so we know when the congestion is on the dhadi channels or if it is an external/operator congestion case. We have servers running both ways, even with combination between ChanisAvail() and the special 130 code. All works very well. PS: ChanisAvail may fail sometimes if you have few available channels and a high demand for incoming calls. Exemple: suppose you have just one channel available, ChanisAvail will return this channel, but before dialplan reaches the Dial() command, an incoming call is received in this channel. So you don't have a channel do dial anymore, but ChanisAvail told you had one. You will then receive a congestion with code 34. Prado -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Benoit Enviada em: domingo, 21 de fevereiro de 2010 2:22 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [asterisk-users] Dahdi Congestion status Hi, I'm using an half T1 line on a asterisk (obviously :)) 1.6.2.4 system, up to recently everything was fine but we are starting to experience the call limitation of the line (15). So as to warn user of the problem i attached a vocal notification to the CONGESTION status after a Dial(), but it looks like it also catch other congestion case (maybe on the receiver side). Should i / Could i use the ChanIsAvail() on a Dahdi channel ? will it work better to detect a full group ? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
