[asterisk-users] DIALSTATUS on CANCEL
Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
[incoming-private]
exten = _X., n, Dial(SIP/1001,30)
exten = _X., n, NoOp(${DIALSTATUS})
exten = _X., n, Gosub(incoming-status,s-${DIALSTATUS},1)
[incoming-status]
exten = s-CANCEL,1, NoOp()
exten = s-CANCEL,n, Return()
exten = s-NOANSWER,1, NoOp()
exten = s-NOANSWER,n, Return()
exten = s-BUSY,1, NoOp()
exten = s-BUSY,n, Return()
This is what we get on a BUSY call:
---
-- Executing [1...@incoming-private:3] Dial(SIP/Proxy-002b,
SIP/1001,50) in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- Got SIP response 486 Busy Here back from 10.0.0.1
-- SIP/1001-002c is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [1...@incoming-private:4] NoOp(SIP/Proxy-002b,
BUSY) in new stack
-- Executing [1...@incoming-private:5] Gosub(SIP/Proxy-002b,
incoming-status,s-BUSY,1) in new stack
This is what we get on a NO ANSWER call:
---
-- Executing [1...@incoming-private:3] Dial(SIP/Proxy-002f,
SIP/1001,30) in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- SIP/1001-0030 is ringing
-- Nobody picked up in 3 ms
-- Executing [1...@incoming-private:4] NoOp(SIP/Proxy-002f,
NOANSWER) in new stack
-- Executing [1...@incoming-private:5] Gosub(SIP/Proxy-002f,
incoming-status,s-NOANSWER,1) in new stack
This is what we get on a CANCEL call:
-
-- Executing [1...@incoming-private:3] Dial(SIP/Proxy-0031,
SIP/1001,30) in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- SIP/1001-0032 is ringing
== Spawn extension (incoming-private, , 3) exited non-zero on
'SIP/Proxy-0031'
There's no event indicating that a DIALSTATUS is generated and the call
simply doesn't go to the next step in the dialplan. Unless I'm missing
something, it seems to me that it might be a bug.
I would be happy to get feedback from other users of the DIALSTATUS value
(or Digium), especially in the CANCEL scenario.
Thank you,
Michael
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[asterisk-users] Asterisk Clustering and DUNDi J Richardson whitepaper
Hi All We're getting to the point where we need to start increasing capacity on our asterisk servers. I've had a look at the DUNDi J Richardson white paper and it seems pretty straight forward. My question is have any of you implemented this solution in a production environment? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Attack problem
Ircd is not installed and cant be located in all system ,any one know or have an idea how do they infect my system, Any bug in asterisknow? How to find the script that initiates this invites ? 135.307281 192.168.138.56 - 218.75.79.17 TCP 36578 ircd [ACK] Seq=36 Ack=111 Win=5840 Len=0 135.307434 192.168.138.56 - 218.75.79.17 TCP 36578 ircd [FIN, ACK] Seq=36 Ack=111 Win=5840 Len=0 135.309188 218.75.79.17 - 192.168.138.56 TCP ircd 36578 [FIN, ACK] Seq=111 Ack=1 Win=4096 Len=0 135.309211 192.168.138.56 - 218.75.79.17 TCP 36578 ircd [ACK] Seq=37 Ack=112 Win=5840 Len=0 135.334037 192.168.138.56 - 192.168.5.2 DNS Standard query A irc3.mysteryaddict.com 135.334496 192.168.5.2 - 192.168.138.56 DNS Standard query response A 87.229.45.226 135.334657 192.168.138.56 - 87.229.45.226 TCP 53718 ircd [SYN] Seq=0 Win=5840 Len=0 MSS=1460 TSV=1532274 TSER=0 WS=7 135.342359 218.75.79.17 - 192.168.138.56 TCP ircd 42802 [SYN, ACK] Seq=0 Ack=1 Win=1460 Len=0 MSS=1380 135.342399 192.168.138.56 - 218.75.79.17 TCP 42802 ircd [ACK] Seq=1 Ack=1 Win=5840 Len=0 135.342554 192.168.138.56 - 218.75.79.17 IRC Request Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Friday, December 17, 2010 6:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Attack problem On Friday 17 Dec 2010, Khaled W. Chehab wrote: HI, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack # /etc/init.d/ircd stop # chmod -x /etc/init.d/ircd Should do the business :) -- AJS -- _ -- 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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
Hi All,
I have some problem with Asterisk 1.8 and DIal() to SIP unreachable friend.
My dialplan:
exten = _,1,Dial(SIP/${EXTEN},60,rt)
Now, when I Dial extension 1050, and there is no 1050 peer registered I got:
[Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len
843) to 0.0.4.26:5060 returned -1: Invalid argument
In 1.6 there was no problem, I have got Channel is UNAVAILABLE message
and hangup.
What have I missed in 1.8?
Regards,
Jarek
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Re: [asterisk-users] Ported Asterisk in Android
On Fri, 17 Dec 2010 15:51:33 +0530, Nikhil d.nik...@cem-solutions.net wrote: Does anyone ported Asterisk to Android OS .please give details www.servalproject.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] start services automatically
Hello All, i have asterisk installed in my call centre without any issue I would like to ask you some questions related to services. i want to start asterisk and httpd and aheevacti automatically when the server centos reboot or shutdown becouse i must start all services manually (service asterisk start ,service httpd start ...) Maybe i must use crontab but I don’t know how to do, any help please Thanks and Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attack problem
netstat -anp |grep 6667 Best Regards, Muhammad Nuzaihan Kamal Network Consultant Mobile: +65 97473874 Asfa Systems Pte Ltd 91, Alps Avenue. #03-10. Singapore 498787 Tel: +65 62538211 Fax: +65 62504814 www.asfasystems.com.sg pub 4096R/36630777 2010-07-10 Key fingerprint = 670A 4D60 0A2D 43A1 2FE0 DFDA D3A9 3F32 3663 0777 uid Muhammad Nuzaihan Kamalluddin (Asfa Systems Pte. Ltd.) muham...@asfasystems.com sub 4096R/97E5CBBD 2010-07-10 On 20-Dec-2010, at 5:40 PM, Khaled W. Chehab wrote: Ircd is not installed and cant be located in all system ,any one know or have an idea how do they infect my system, Any bug in asterisknow? How to find the script that initiates this invites ? 135.307281 192.168.138.56 - 218.75.79.17 TCP 36578 ircd [ACK] Seq=36 Ack=111 Win=5840 Len=0 135.307434 192.168.138.56 - 218.75.79.17 TCP 36578 ircd [FIN, ACK] Seq=36 Ack=111 Win=5840 Len=0 135.309188 218.75.79.17 - 192.168.138.56 TCP ircd 36578 [FIN, ACK] Seq=111 Ack=1 Win=4096 Len=0 135.309211 192.168.138.56 - 218.75.79.17 TCP 36578 ircd [ACK] Seq=37 Ack=112 Win=5840 Len=0 135.334037 192.168.138.56 - 192.168.5.2 DNS Standard query A irc3.mysteryaddict.com 135.334496 192.168.5.2 - 192.168.138.56 DNS Standard query response A 87.229.45.226 135.334657 192.168.138.56 - 87.229.45.226 TCP 53718 ircd [SYN] Seq=0 Win=5840 Len=0 MSS=1460 TSV=1532274 TSER=0 WS=7 135.342359 218.75.79.17 - 192.168.138.56 TCP ircd 42802 [SYN, ACK] Seq=0 Ack=1 Win=1460 Len=0 MSS=1380 135.342399 192.168.138.56 - 218.75.79.17 TCP 42802 ircd [ACK] Seq=1 Ack=1 Win=5840 Len=0 135.342554 192.168.138.56 - 218.75.79.17 IRC Request Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Friday, December 17, 2010 6:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Attack problem On Friday 17 Dec 2010, Khaled W. Chehab wrote: HI, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack # /etc/init.d/ircd stop # chmod -x /etc/init.d/ircd Should do the business :) -- AJS -- _ -- 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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- _ -- 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] dahdi show channels / how to get the call duration for active calls?
So simple - great, thank!!! Am 17.12.2010 13:07, schrieb Vincius Fontes: You probably want "core show channels verbose". Atenciosamente, Vincius Fontes Gerente de Segurana da Informao Canall Tecnologia em Comunicaes Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicaes Passo Fundo - RS - Brazil +55 54 2104-7000 Hi, for dahdi-calls I can see the current calls with "dahdi show channels". But where can I see the current call-duration or the call-start-time? "dahdi show channel n" does not show this info. -Thorsten- -- _ -- 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 -- Thorsten Gllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Dsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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] dahdi show channels / how to get the call duration for active calls?
So simple - great, thank!!! Am 17.12.2010 13:07, schrieb Vincius Fontes: You probably want "core show channels verbose". Atenciosamente, Vincius Fontes Gerente de Segurana da Informao Canall Tecnologia em Comunicaes Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicaes Passo Fundo - RS - Brazil +55 54 2104-7000 Hi, for dahdi-calls I can see the current calls with "dahdi show channels". But where can I see the current call-duration or the call-start-time? "dahdi show channel n" does not show this info. -Thorsten- -- _ -- 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 -- Thorsten Gllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Dsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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] Pass DTMF to IVR gateway through SIP phone conferencing.
Will someone help/direct me find a way to implement this? Or you can suggest some other method. On Fri, Dec 17, 2010 at 12:44 PM, Asterisk Man theasterisk...@gmail.comwrote: Hi friends, I want to implement following scenario using Asterisk. Please suggest me whether it is possible or not. This is bit off Asterisk and more on SIP side. An Asterisk box with one Station(SIP channel) and PRI. Agent dials a PSTN number of customer from station through Asterisk PRI. Agent gets connected with customer. Agent puts customer on hold. Agent dials another PSTN number which is of IVR gateway. Agent now makes conference(Station facility) with customer and IVR gateway. Gateway plays an IVR asking customer to enter his customer id number. My question is, will DTMF get forwarded to IVR gateway? I am asked to implement this and not having PRI for the moment in my Asterisk box. Thanking you in advance. -AsteriskMan -- _ -- 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] Ported Asterisk in Android
i believe there is a way to do it using asterisk and flashphoner ++ 2010/12/20 Gilles codecompl...@free.fr On Fri, 17 Dec 2010 15:51:33 +0530, Nikhil d.nik...@cem-solutions.net wrote: Does anyone ported Asterisk to Android OS .please give details www.servalproject.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cordialement Gabriel 09 79 94 71 13 -- _ -- 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] start services automatically
When installing asterisk you should type make config to have asterisk create init script automatically. For http chkconfig httpd on Adolphe Cher-aime From my Iphone On Dec 20, 2010, at 5:48 AM, salaheddine elharit salah.elharit...@gmail.com wrote: Hello All, i have asterisk installed in my call centre without any issue I would like to ask you some questions related to services. i want to start asterisk and httpd and aheevacti automatically when the server centos reboot or shutdown becouse i must start all services manually (service asterisk start ,service httpd start ...) Maybe i must use crontab but I don’t know how to do, any help please Thanks and Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] start services automatically
salaheddine elharit wrote: becouse i must start all services manually (service asterisk start ,service httpd start chkconfig httpd on chkconfig asterisk on 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
[asterisk-users] dahdi issue on digium AEX800
Hi All, I have installed asterisk 1.6.2.8 Dahdi: 2.4.0 Digium card: Digium, Inc. Wildcard AEX800 8-port analog card I have configured this card properly and it is working for calls too, But there is issue of one way audio on this outbound routes only, My voice is going to outside pstn number but i am not able to get their audio, I have disabled firewall, selinux is also off. If I am applying this line to analog phone then also it is working fine, But when it is added on digium card then this issue happens, can anybody help me for this issue? Thanks, Max Alex Voip Developer -- _ -- 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] In which version is eventfilter working?
On 10-12-19 06:23 AM, Daniel Knoll wrote: In which Version of Asterisk is EventFilter: in manager.conf working? Higher than 1.6.2.10 or from the 1.8.0 Version? Always refer to CHANGES[1] or UPGRADE.txt. It was added in 1.8 [1] http://svn.digium.com/svn/asterisk/branches/1.8/CHANGES -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pass DTMF to IVR gateway through SIP phone conferencing.
Without reading too much into your description, I can tell you that being an inband sound, and as long as the dtmf tone is heard by everybody during the conference, and being the ivr gateway one of the parties of the conference, I don't see a reason why the ivr gateway wouldn't act upon hearing the dtmf tone. It wouldn't know who pressed it, although if that matters, can be arranged by writing a patch to the meetme application where you can identify the channel that pressed the dtmf tone. Best Chris Savinovich On December 20, 2010 at 6:56 AM Asterisk Man theasterisk...@gmail.com wrote: Will someone help/direct me find a way to implement this? Or you can suggest some other method. On Fri, Dec 17, 2010 at 12:44 PM, Asterisk Man theasterisk...@gmail.com [mailto:theasterisk...@gmail.com] wrote: Hi friends, I want to implement following scenario using Asterisk. Please suggest me whether it is possible or not. This is bit off Asterisk and more on SIP side. An Asterisk box with one Station(SIP channel) and PRI. Agent dials a PSTN number of customer from station through Asterisk PRI. Agent gets connected with customer. Agent puts customer on hold. Agent dials another PSTN number which is of IVR gateway. Agent now makes conference(Station facility) with customer and IVR gateway. Gateway plays an IVR asking customer to enter his customer id number. My question is, will DTMF get forwarded to IVR gateway? I am asked to implement this and not having PRI for the moment in my Asterisk box. Thanking you in advance. -AsteriskMan Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi issue on digium AEX800
On 12/20/10 6:50 AM, Max Alex wrote: Hi All, I have installed asterisk 1.6.2.8 Dahdi: 2.4.0 Digium card: Digium, Inc. Wildcard AEX800 8-port analog card I have configured this card properly and it is working for calls too, But there is issue of one way audio on this outbound routes only, My voice is going to outside pstn number but i am not able to get their audio, I have disabled firewall, selinux is also off. First off, my recommendation is for you to contact Digium technical support. They will be able to help you triage and resolve this. That being said, some questions that come to mind looking at your description: 1) Is the one-way audio constant or intermittent? Does it affect all calls through the card or only outbound routes? 2) Do you have a VPM module installed on the card? If so, does loading the card with modprobe wctdm24xxp vpmsupport=0 change the behavior at all? 3) Is there anything strange you see in the output of the dmesg command? 4) Are you able to install and try the current trunk of DAHDI? Does that change what you see? Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] start services automatically
ok thank you so much for your help 2010/12/20 Doug Lytle supp...@drdos.info salaheddine elharit wrote: becouse i must start all services manually (service asterisk start ,service httpd start chkconfig httpd on chkconfig asterisk on 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 -- _ -- 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 and Alcatel digital phone's
Hi Jonathan, I already looked at their product a few weeks ago, but because Alcatel wasn't on their list of compatible devices, I left it alone. Because of your email, I went looking on their site for a second time and noticed on their blog that they're experimenting with Alcatel devices. So after emailing them, there is a chance that we could use their product for our digital Alcatel phones. So fingers crossed and thanks for the info ;) Sander -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Jonathan C. Bailey Verzonden: zaterdag 18 december 2010 18:19 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Asterisk and Alcatel digital phone's There is a product from Citel (the TVA) that we're currently using with Toshiba phones. I know they also support Avaya, Nortel, and Panasonic, but am not sure if they do any other brands. They more or less convert your old digital phones to SIP. They have have full compatibility information on their website... -Jon - Original Message - From: John Novack jnov...@stromberg-carlson.org To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 18, 2010 6:48:57 AM Subject: Re: [asterisk-users] Asterisk and Alcatel digital phone's Sander Naudts wrote: Asterisk and Alcatel digital phone's Hi, I'm sorry if this is already asked somewhere on the list but I couldn't find it. We have an old PBX system controlled by our Telecom provider. There are analog phones but also digital alcatel phone's connected to it. These are not ip based but legacy digital phone's. Is there a way how we can connect them to our own Asterisk PBX? The old PBX is going to be removed, so it has to be a solution: Digital alacatel phone - directly connected to Asterisk. Short answer NO! What you are calling legacy digital phones are not universal, and for many years have been integrated with the host system. This is generally true for business systems from 2 lines and six stations to large systems with hundreds of phones. Is there some hardware gateway or something we can use? The only gatewaywill be your existing switch or another of the same generation. When the switch is removed, why would the phones not be? the analog phones, if they are not special, but POTS phones that could be used anywhere on a loop start line in a business or home could be reused, but you may find that you will not want to. We looked at the Grandstream GXW4024 gateway for our analog phones but I'm not sure the digital one's can connect to that one as well. No they cannot. Better plan on replacing all the Alcatel phones with IP ones. John Novack Kind regards, Sander Naudts -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] start services automatically
when i make chkconfig httpd on and chkconfig asterisk on with chkconfig --list i found httpd 0:off 1:off 2:on3:on4:on5:on6:off asterisk0:off 1:off 2:on3:on4:on5:on6:off the 0,1, and 6 are OFF just the 2,3,4,5 are ON , and when i reboot the server i found that the service httpd is off with command service httpd status and service asterisk status please advice Best Regards, 2010/12/20 salaheddine elharit salah.elharit...@gmail.com ok thank you so much for your help 2010/12/20 Doug Lytle supp...@drdos.info salaheddine elharit wrote: becouse i must start all services manually (service asterisk start ,service httpd start chkconfig httpd on chkconfig asterisk on 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 -- _ -- 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] Recommendation for a Linux based SCADA
I'm not certain what you mean by needing to setup up a SCADA solution? I assume you want to connect an industrial data acquisition and control system to Asterisk. We have a SCADA system interfaced with Asterisk in our facility. The SCADA hardware we use is the SNAP PAC system from Opto22http://www.opto22.comwhich provides a linux SDK http://www.opto22.com/site/downloads/dl_drilldown.aspx?aid=2890 . You also can set the Opto hardware to send SNMP messages on certain conditions. On Wed, Dec 15, 2010 at 4:23 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous experience in this type of monitoring and automization. I'll be using SNMP data from asterisk servers and endpoints. If anybody has any suggestion which SCADA software can fit in such a VoIP solution, your guidance will be highly appreciated. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.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 -- Marvin Horst -- _ -- 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 and Alcatel digital phone's
No problem. We've had good luck with them so far. Support is also VERY responsive (had a work around in a few hours, and a firmware upgrade to fix the issue within a day or two). -Jon - Original Message - From: Sander Naudts s.nau...@intersui.be To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 20, 2010 8:17:18 AM Subject: Re: [asterisk-users] Asterisk and Alcatel digital phone's Hi Jonathan, I already looked at their product a few weeks ago, but because Alcatel wasn't on their list of compatible devices, I left it alone. Because of your email, I went looking on their site for a second time and noticed on their blog that they're experimenting with Alcatel devices. So after emailing them, there is a chance that we could use their product for our digital Alcatel phones. So fingers crossed and thanks for the info ;) Sander -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Jonathan C. Bailey Verzonden: zaterdag 18 december 2010 18:19 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Asterisk and Alcatel digital phone's There is a product from Citel (the TVA) that we're currently using with Toshiba phones. I know they also support Avaya, Nortel, and Panasonic, but am not sure if they do any other brands. They more or less convert your old digital phones to SIP. They have have full compatibility information on their website... -Jon - Original Message - From: John Novack jnov...@stromberg-carlson.org To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 18, 2010 6:48:57 AM Subject: Re: [asterisk-users] Asterisk and Alcatel digital phone's Sander Naudts wrote: Asterisk and Alcatel digital phone's Hi, I'm sorry if this is already asked somewhere on the list but I couldn't find it. We have an old PBX system controlled by our Telecom provider. There are analog phones but also digital alcatel phone's connected to it. These are not ip based but legacy digital phone's. Is there a way how we can connect them to our own Asterisk PBX? The old PBX is going to be removed, so it has to be a solution: Digital alacatel phone - directly connected to Asterisk. Short answer NO! What you are calling legacy digital phones are not universal, and for many years have been integrated with the host system. This is generally true for business systems from 2 lines and six stations to large systems with hundreds of phones. Is there some hardware gateway or something we can use? The only gatewaywill be your existing switch or another of the same generation. When the switch is removed, why would the phones not be? the analog phones, if they are not special, but POTS phones that could be used anywhere on a loop start line in a business or home could be reused, but you may find that you will not want to. We looked at the Grandstream GXW4024 gateway for our analog phones but I'm not sure the digital one's can connect to that one as well. No they cannot. Better plan on replacing all the Alcatel phones with IP ones. John Novack Kind regards, Sander Naudts -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] dahdi issue on digium AEX800
Am 20.12.2010 15:11, schrieb Shaun Ruffell: On 12/20/10 6:50 AM, Max Alex wrote: Hi All, I have installed asterisk 1.6.2.8 Dahdi: 2.4.0 Digium card: Digium, Inc. Wildcard AEX800 8-port analog card I have configured this card properly and it is working for calls too, But there is issue of one way audio on this outbound routes only, My voice is going to outside pstn number but i am not able to get their audio, I have disabled firewall, selinux is also off. First off, my recommendation is for you to contact Digium technical support. They will be able to help you triage and resolve this. That being said, some questions that come to mind looking at your description: 1) Is the one-way audio constant or intermittent? Does it affect all calls through the card or only outbound routes? 2) Do you have a VPM module installed on the card? If so, does loading the card with modprobe wctdm24xxp vpmsupport=0 change the behavior at all? 3) Is there anything strange you see in the output of the dmesg command? 4) Are you able to install and try the current trunk of DAHDI? Does that change what you see? Cheers, Shaun Please show us -- _ -- 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] dahdi issue on digium AEX800
Am 20.12.2010 15:11, schrieb Shaun Ruffell: On 12/20/10 6:50 AM, Max Alex wrote: Hi All, I have installed asterisk 1.6.2.8 Dahdi: 2.4.0 Digium card: Digium, Inc. Wildcard AEX800 8-port analog card I have configured this card properly and it is working for calls too, But there is issue of one way audio on this outbound routes only, My voice is going to outside pstn number but i am not able to get their audio, I have disabled firewall, selinux is also off. First off, my recommendation is for you to contact Digium technical support. They will be able to help you triage and resolve this. That being said, some questions that come to mind looking at your description: 1) Is the one-way audio constant or intermittent? Does it affect all calls through the card or only outbound routes? 2) Do you have a VPM module installed on the card? If so, does loading the card with modprobe wctdm24xxp vpmsupport=0 change the behavior at all? 3) Is there anything strange you see in the output of the dmesg command? 4) Are you able to install and try the current trunk of DAHDI? Does that change what you see? Cheers, Shaun Please show us cat /etc/asterisk/chan_dahdi.conf | grep overlapdial is it yes? -- _ -- 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] dahdi issue on digium AEX800
Am 20.12.2010 16:00, schrieb Thorsten Göllner: Am 20.12.2010 15:11, schrieb Shaun Ruffell: On 12/20/10 6:50 AM, Max Alex wrote: Hi All, I have installed asterisk 1.6.2.8 Dahdi: 2.4.0 Digium card: Digium, Inc. Wildcard AEX800 8-port analog card I have configured this card properly and it is working for calls too, But there is issue of one way audio on this outbound routes only, My voice is going to outside pstn number but i am not able to get their audio, I have disabled firewall, selinux is also off. First off, my recommendation is for you to contact Digium technical support. They will be able to help you triage and resolve this. That being said, some questions that come to mind looking at your description: 1) Is the one-way audio constant or intermittent? Does it affect all calls through the card or only outbound routes? 2) Do you have a VPM module installed on the card? If so, does loading the card with modprobe wctdm24xxp vpmsupport=0 change the behavior at all? 3) Is there anything strange you see in the output of the dmesg command? 4) Are you able to install and try the current trunk of DAHDI? Does that change what you see? Cheers, Shaun Please show us Ups, last line was cut. Please show us: cat /etc/asterisk/chan_dahdi.conf | grep overlapdial -- _ -- 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] start services automatically
On Mon, Dec 20, 2010 at 02:34:23PM +, salaheddine elharit wrote: the 0,1, and 6 are OFF just the 2,3,4,5 are ON , and when i reboot the server i found that the service httpd is off with command service httpd status and service asterisk status please advice This is just one of many problems you will encounter. You need to train or hire an actual Unix/Linux system administrator. -- _ -- 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] Call sip:u...@domain.com?
On Fri, 17 Dec 2010 17:54:00 +, Roger Burton West ro...@firedrake.org wrote: How would you _expect_ to be able to specify a destination server from a telephone keypad? Thanks guys for the infos. My goal was to learn how to configure Asterisk so it could call SIP URI (u...@domain) using XLite, but didn't consider the issue of regular phones, which only have a keypad. I'll read up about Freenum, ENUM/E164, SIPBroker etc. to learn how to map a SIP URI to a digit-only number. Thank you. -- _ -- 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 1.8 - Dial problem if SIP friend is UNREACHABLE.
On 12/20/2010 4:41 AM, Jarek Jarzebowski wrote: Now, when I Dial extension 1050, and there is no 1050 peer registered I got: [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len 843) to 0.0.4.26:5060 returned -1: Invalid argument You haven't done anything wrong; I have the same issue. Just add it to the list of things to fix in 1.8.. Do you want to add it to http://issues.asterisk.org ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
On 10-12-20 04:41 AM, Jarek Jarzebowski wrote: [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len 843) to 0.0.4.26:5060 returned -1: Invalid argument It looks to be a regression with the IPv6 code added to chan_sip. Which version of 1.8 are you using? I'd also be good to see a full debug[1] log. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting `userfield` from within a callfile
Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application (written by someone else before me) which sets up calls by creating files of the general form Channel: SIP/$INSIDE_NUMBER Context: $CONTEXT Extension: $OUTSIDE_NUMBER Priority: 1 CallerId: $INSIDE_NUMBER in /var/spool/asterisk/outgoing/ . It works very well. However, it would be nice to be able to attach an additional piece of information along with the call record There is a userfield in the SQL database, which is a VARCHAR(255) and would be plenty for what we need. Is there a way to set the userfield of the CDR database from within such a callfile? -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Anyone going to remove this spammer/scammer? 2010/12/19 Dmitry Kupchinetsky dkupchinet...@hotmail.com: http://www.barenakedbabies.com/shop/images/images.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unexpected dialplan match
I was wondering why *...@default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI dialplan show *...@default
'_*[0-9a-zA-Z].*0.' =
1. NoOp(${EXTEN}) [pbx_config]
2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config]
3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config]
4. Set(CDR(accountcode)=${accountcode}) [pbx_config]
7. ResetCDR() [pbx_config]
8. ...
--
Daniel Tryba
--
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[asterisk-users] SIP 420
Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it’s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? --- SIP read from UDP://x.x.x.x:5060 http://10.247.1.126:5060 --- INVITE sip:4...@x.x.x.x:5060;transport=udpsip:4...@s144701.trixbox.fonality.com:5060;transport=udp SIP/2.0 To: sip:4...@x.x.x.x5060;transport=udpsip:4...@s144701.trixbox.fonality.com:5060;transport=udp From: sip:3...@x.x.x.x:5060http://sip:3...@10.247.1.126:5060 ;tag=4f5cb549 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;rport Call-ID: 350da2493d160...@zhqtzgvsbdn2zhiwzjeuqu0usuruq09suc5orvq. CSeq: 1 INVITE Contact: sip:3...@x.x.x.x:5060http://sip:3...@10.247.1.126:5060 Max-Forwards: 70 Session-Expires: 1800 Min-SE: 90 Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY Content-Type: application/sdp *Require: x-call-detail* Supported: timer User-Agent: xxx PBX Phone 1.1.0.10434 ORIG/IDCB01S110 SN/001d09048211 (Windows NT 5.1) Content-Length: 426 v=0 o=SIP 1292608808 1292608808 IN IP4 x.x.x.x s=SIP c=IN IP4 x.x.x.x t=1292608808 0 m=audio 1 RTP/AVP 97 103 100 127 0 8 102 18 4 101 a=rtpmap:97 IPCMWB/16000 a=rtpmap:103 ISAC/16000 a=rtpmap:100 EG711U/8000 a=rtpmap:127 EG711A/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:102 iLBC/8000 a=fmtp:102 mode=30 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 - --- (17 headers 17 lines) --- == Using SIP RTP CoS mark 5 --- Transmitting (no NAT) to x.x.x.x:5060 http://10.247.1.126:5060 --- SIP/2.0 420 Bad extension (unsupported) Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;received=x.x.x.x;rport=5060 From: sip:3...@x.x.x.x:5060http://sip:3...@10.247.1.126:5060 ;tag=4f5cb549 To: sip:4...@x.x.x.x:5060;transport=udpsip:4...@s144701.trixbox.fonality.com:5060;transport=udp ;tag=as34f3ff9f Call-ID: 350da2493d160...@zhqtzgvsbdn2zhiwzjeuqu0usuruq09suc5orvq. CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.28 llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Date: Fri, 17 Dec 2010 18:00:04 GMT *Unsupported: x-call-detail* Content-Length: 0 --Dovey Forman -- _ -- 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] Setting `userfield` from within a callfile
2010/12/20 A J Stiles asterisk_l...@earthshod.co.uk Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application (written by someone else before me) which sets up calls by creating files of the general form Channel: SIP/$INSIDE_NUMBER Context: $CONTEXT Extension: $OUTSIDE_NUMBER Priority: 1 CallerId: $INSIDE_NUMBER in /var/spool/asterisk/outgoing/ . It works very well. However, it would be nice to be able to attach an additional piece of information along with the call record There is a userfield in the SQL database, which is a VARCHAR(255) and would be plenty for what we need. Is there a way to set the userfield of the CDR database from within such a callfile? Yes, adding a Set field in your call file (see http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out), you'll be able to pass everything you need to your dialplan, and then, from there, write everything you need to your CDR. Cheers -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting `userfield` from within a callfile
On Monday 20 December 2010 10:33:33 A J Stiles wrote: Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application (written by someone else before me) which sets up calls by creating files of the general form Channel: SIP/$INSIDE_NUMBER Context: $CONTEXT Extension: $OUTSIDE_NUMBER Priority: 1 CallerId: $INSIDE_NUMBER in /var/spool/asterisk/outgoing/ . It works very well. However, it would be nice to be able to attach an additional piece of information along with the call record There is a userfield in the SQL database, which is a VARCHAR(255) and would be plenty for what we need. Is there a way to set the userfield of the CDR database from within such a callfile? As is stated within sample.call (in the root directory of the Asterisk source): # # You can set channel variables that will be passed to the channel. # This includes writable dialplan functions. To set a writable dialplan # function, the module containing this function *must* be loaded. # #Set: file1=/tmp/to #Set: file2=/tmp/msg #Set: timestamp=20021023104500 #Set: CDR(userfield,r)=42 -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unexpected dialplan match
On 12/20/2010 11:35 AM, Daniel Tryba wrote:
I was wondering why *...@default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI dialplan show *...@default
'_*[0-9a-zA-Z].*0.' =
1. NoOp(${EXTEN}) [pbx_config]
2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config]
3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config]
4. Set(CDR(accountcode)=${accountcode}) [pbx_config]
7. ResetCDR() [pbx_config]
8. ...
'.' stops further matching. Your extension ends up being (effectively)
shortened to _*[0-9a-zA-Z].
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Re: [asterisk-users] Call sip:u...@domain.com?
On 12/17/2010 06:25 AM, Gilles wrote: On Thu, 16 Dec 2010 11:54:31 +0100, Gillescodecompl...@free.fr wrote: Now, I'd like to be able to call any number on the Net that is advertised as sip:u...@domain.com, such as those: I mean: Do I really have to first create a section in sip.conf each time a user needs to call a number on a new SIP server? http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net You've missed a very important point here: you are using a *SIP* endpoint to call a *SIP* URI. The endpoint can do that directly, and doesn't need any help from Asterisk to do it. If you wanted to be able to restrict/control such calls, you'd need to use a SIP proxy... but Asterisk is not a proxy. Asterisk is a Back-to-Back User Agent, which means whatever URI the endpoint sends to Asterisk terminates there, and Asterisk constructs an outbound URI of some form, connecting the two channels together. You should probably take a step back and ask yourself what value Asterisk would bring being in the middle between your SIP softphones and some random SIP endpoint out on the Internet. Once you determine that, you'll know whether it's worth trying to construct a solution for this or not. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendation for a Linux based SCADA
Thanks for this info. It seems like good hardware and software solution provider. I'll explore it a bit more and see if it fits my client's need. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com On 2010-12-20 9:56 AM, marvin horst fivehor...@gmail.com wrote: I'm not certain what you mean by needing to setup up a SCADA solution? I assume you want to connect an industrial data acquisition and control system to Asterisk. We have a SCADA system interfaced with Asterisk in our facility. The SCADA hardware we use is the SNAP PAC system from Opto22http://www.opto22.comwhich provides a linux SDK http://www.opto22.com/site/downloads/dl_drilldown.aspx?aid=2890 . You also can set the Opto hardware to send SNMP messages on certain conditions. On Wed, Dec 15, 2010 at 4:23 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous e... -- _ -- 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 -- Marvin Horst -- _ -- 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] SIP 420
On Mon, Dec 20, 2010 at 9:46 AM, Dovey Forman dovey.for...@idt.net wrote: I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it’s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? Yes. The softphone is requiring x-call-detail, which Asterisk does not support. The softphone either needs to drop that requirement completely, or change it to a Supported header so it can be processed by other SIP servers. -Jonathan -- _ -- 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] SIP 420
On 12/20/2010 11:46 AM, Dovey Forman wrote: Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it’s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? --- SIP read from UDP://x.x.x.x:5060 http://10.247.1.126:5060--- INVITEsip:4...@x.x.x.x:5060;transport=udp sip:4...@s144701.trixbox.fonality.com:5060;transport=udpSIP/2.0 To: sip:4...@x.x.x.x5060;transport=udp sip:4...@s144701.trixbox.fonality.com:5060;transport=udp From: sip:3...@x.x.x.x:5060 http://sip:3...@10.247.1.126:5060;tag=4f5cb549 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;rport Call-ID: 350da2493d160...@zhqtzgvsbdn2zhiwzjeuqu0usuruq09suc5orvq. CSeq: 1 INVITE Contact: sip:3...@x.x.x.x:5060 http://sip:3...@10.247.1.126:5060 Max-Forwards: 70 Session-Expires: 1800 Min-SE: 90 Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY Content-Type: application/sdp *Require: x-call-detail* Supported: timer User-Agent: xxx PBX Phone 1.1.0.10434 ORIG/IDCB01S110 SN/001d09048211 (Windows NT 5.1) Content-Length: 426 v=0 o=SIP 1292608808 1292608808 IN IP4 x.x.x.x s=SIP c=IN IP4 x.x.x.x t=1292608808 0 m=audio 1 RTP/AVP 97 103 100 127 0 8 102 18 4 101 a=rtpmap:97 IPCMWB/16000 a=rtpmap:103 ISAC/16000 a=rtpmap:100 EG711U/8000 a=rtpmap:127 EG711A/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:102 iLBC/8000 a=fmtp:102 mode=30 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 - --- (17 headers 17 lines) --- == Using SIP RTP CoS mark 5 --- Transmitting (no NAT) tox.x.x.x:5060 http://10.247.1.126:5060--- SIP/2.0 420 Bad extension (unsupported) Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;received=x.x.x.x;rport=5060 From: sip:3...@x.x.x.x:5060 http://sip:3...@10.247.1.126:5060;tag=4f5cb549 To: sip:4...@x.x.x.x:5060;transport=udp sip:4...@s144701.trixbox.fonality.com:5060;transport=udp;tag=as34f3ff9f Call-ID: 350da2493d160...@zhqtzgvsbdn2zhiwzjeuqu0usuruq09suc5orvq. CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.28 llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Date: Fri, 17 Dec 2010 18:00:04 GMT *Unsupported: x-call-detail* Content-Length: 0 This is pretty clear... your softphone is requiring support for a private SIP extension called 'call-detail', and since Asterisk does not support it, it cannot process the INVITE request. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unexpected dialplan match
On Monday 20 December 2010 11:35:21 Daniel Tryba wrote:
I was wondering why *...@default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI dialplan show *...@default
'_*[0-9a-zA-Z].*0.' =
1. NoOp(${EXTEN}) [pbx_config]
2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config]
3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config]
4. Set(CDR(accountcode)=${accountcode}) [pbx_config]
7. ResetCDR() [pbx_config]
8. ...
You. . is a short-circuit operator; everything after it is ignored.
--
Tilghman
--
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Re: [asterisk-users] Recommendation for a Linux based SCADA
Any device that you can talk to and be used in Linux can be interfaced into asterisk with the power of AGI. I have some WebRelay modules that I can remotely control via asterisk. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Monday, December 20, 2010 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Recommendation for a Linux based SCADA Thanks for this info. It seems like good hardware and software solution provider. I'll explore it a bit more and see if it fits my client's need. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com On 2010-12-20 9:56 AM, marvin horst fivehor...@gmail.com wrote: I'm not certain what you mean by needing to setup up a SCADA solution? I assume you want to connect an industrial data acquisition and control system to Asterisk. We have a SCADA system interfaced with Asterisk in our facility. The SCADA hardware we use is the SNAP PAC system from Opto22 http://www.opto22.com which provides a linux SDK http://www.opto22.com/site/downloads/dl_drilldown.aspx?aid=2890 . You also can set the Opto hardware to send SNMP messages on certain conditions. On Wed, Dec 15, 2010 at 4:23 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous e... -- _ -- 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 -- Marvin Horst -- _ -- 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] Asterisk 1.6 produces *many* zombie processes on Debian.
We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. -- _ -- 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] Unexpected dialplan match
On Mon, Dec 20, 2010 at 12:27:46PM -0600, Jason Parker wrote: '_*[0-9a-zA-Z].*0.' '.' stops further matching. Your extension ends up being (effectively) shortened to _*[0-9a-zA-Z]. That explains a lot, never read it this way before. Thanks for the eye opener. What I'm looking for is a extension that handles: ^\*\w+\*\d+$ I guess I'll have to catch _*. and manually check if it matches above regexp. -- 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
[asterisk-users] Upgrading DAHDI and Asterisk
Hello, I just upgraded DAHDI from 2.3.0 to 2.4.0, both dahdi-linux and dahdi-tools (Ubuntu 10.04 64 bits). It took a few minutes, and it was straightforward. Everything is working. Now I'm preparing to upgrade Asterisk from 1.6.2.7 to 1.6.2.15. I made a backup of configuration files, codec licenses and CDR. Is there something else I should be aware of before upgrading? Thank you, Alex Saavedra -- _ -- 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] cdr_mysql stopped working
I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql table for CDR's today there are no entries since the update. I have rebuilt and re-installed and re-started asterisk still no CDR's flowing to mysql. I did not change any configs. I checked to make sure that the cdr_mysql option was selected under the make menu options. The module shows it is there when I do a modules show. I don't get any errors saying it can't write to the table. My voicemail settings are pulling from the same server. Any ideas on what I could try to fix this or how I could test to see what is causing it? Thanks Bryant -- _ -- 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] What's up?
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Re: [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
2010/12/20 Jeremy Kister asterisk...@jeremykister.com: On 12/20/2010 4:41 AM, Jarek Jarzebowski wrote: Now, when I Dial extension 1050, and there is no 1050 peer registered I got: [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len 843) to 0.0.4.26:5060 returned -1: Invalid argument You haven't done anything wrong; I have the same issue. Just add it to the list of things to fix in 1.8.. Do you want to add it to http://issues.asterisk.org ? Yes, of course. I want to add it to http://issues.asterisk.org -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jarek -- _ -- 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 1.8 - Dial problem if SIP friend is UNREACHABLE.
2010/12/20 Paul Belanger pabelan...@digium.com: On 10-12-20 04:41 AM, Jarek Jarzebowski wrote: [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len 843) to 0.0.4.26:5060 returned -1: Invalid argument It looks to be a regression with the IPv6 code added to chan_sip. Which version of 1.8 are you using? I'd also be good to see a full debug[1] log. OK, so I have attached debug log. I am using: *CLI core show version Asterisk 1.8.1.1 built by root @ asterisk on a i686 running Linux on 2010-12-17 23:03:58 UTC [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Jarek full.tgz Description: GNU Zip compressed data -- _ -- 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 1.8 - Dial problem if SIP friend is UNREACHABLE.
On 10-12-20 05:51 PM, Jarek Jarzebowski wrote: OK, so I have attached debug log. I am using: *CLI core show version Asterisk 1.8.1.1 built by root @ asterisk on a i686 running Linux on 2010-12-17 23:03:58 UTC Definitely a bug, ran into the same issue with chan_iax2 and DNS lookups. Please open a new issue on the tracker, include your debug log and sip.conf. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 420
Thanks Kevin. Did it work with Asterisk 1.2 because it ignored it? Why now? On Dec 20, 2010 3:28 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/20/2010 11:46 AM, Dovey Forman wrote: Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it’s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? --- SIP read from UDP://x.x.x.x:5060 http://10.247.1.126:5060--- INVITEsip:4...@x.x.x.x:5060;transport=udp sip:4...@s144701.trixbox.fonality.com:5060;transport=udpSIP/2.0 To: sip:4...@x.x.x.x5060;transport=udp sip:4...@s144701.trixbox.fonality.com:5060;transport=udp From: sip:3...@x.x.x.x:5060 http://sip:3...@10.247.1.126:5060;tag=4f5cb549 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;rport Call-ID: 350da2493d160...@zhqtzgvsbdn2zhiwzjeuqu0usuruq09suc5orvq. CSeq: 1 INVITE Contact: sip:3...@x.x.x.x:5060 http://sip:3...@10.247.1.126:5060 Max-Forwards: 70 Session-Expires: 1800 Min-SE: 90 Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY Content-Type: application/sdp *Require: x-call-detail* Supported: timer User-Agent: xxx PBX Phone 1.1.0.10434 ORIG/IDCB01S110 SN/001d09048211 (Windows NT 5.1) Content-Length: 426 v=0 o=SIP 1292608808 1292608808 IN IP4 x.x.x.x s=SIP c=IN IP4 x.x.x.x t=1292608808 0 m=audio 1 RTP/AVP 97 103 100 127 0 8 102 18 4 101 a=rtpmap:97 IPCMWB/16000 a=rtpmap:103 ISAC/16000 a=rtpmap:100 EG711U/8000 a=rtpmap:127 EG711A/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:102 iLBC/8000 a=fmtp:102 mode=30 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 - --- (17 headers 17 lines) --- == Using SIP RTP CoS mark 5 --- Transmitting (no NAT) tox.x.x.x:5060 http://10.247.1.126:5060--- SIP/2.0 420 Bad extension (unsupported) Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;received=x.x.x.x;rport=5060 From: sip:3...@x.x.x.x:5060 http://sip:3...@10.247.1.126:5060;tag=4f5cb549 To: sip:4...@x.x.x.x:5060;transport=udp sip:4...@s144701.trixbox.fonality.com:5060 ;transport=udp;tag=as34f3ff9f Call-ID: 350da2493d160...@zhqtzgvsbdn2zhiwzjeuqu0usuruq09suc5orvq. CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.28 llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Date: Fri, 17 Dec 2010 18:00:04 GMT *Unsupported: x-call-detail* Content-Length: 0 This is pretty clear... your softphone is requiring support for a private SIP extension called 'call-detail', and since Asterisk does not support it, it cannot process the INVITE request. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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
