[asterisk-users] (no subject)
Hi I am running following asterisk installed with apt on Debian 7.1. dpkg -l |grep asterisk ii asterisk 1:1.8.13.1~dfsg-3+deb7u1 amd64Open Source Private Branch Exchange (PBX) ii asterisk-config1:1.8.13.1~dfsg-3+deb7u1 all Configuration files for Asterisk ii asterisk-core-sounds-en-gsm1.4.22-1 all asterisk PBX sound files - en-us/gsm ii asterisk-modules 1:1.8.13.1~dfsg-3+deb7u1 amd64loadable modules for the Asterisk PBX If the incoming INVITE has the following two multiple bodies then it would not respond to that. It won't even send a Trying. We are using* TCP *only. Content-Type: application/sdp Content-Type: application/ISUP;base=itu-t92+;version=itu-t92+. Is this is a known issue? Are later version of asterisk able to deal with such multi-bodies INVITE? I got to play early media so it needs to make some sense out of first SDP. Best regards, Adnan -- _ -- 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] Dealing with muti-body INVITE
Hi I am running following asterisk installed with apt on Debian 7.1. dpkg -l |grep asterisk ii asterisk 1:1.8.13.1~dfsg-3+deb7u1 amd64Open Source Private Branch Exchange (PBX) ii asterisk-config1:1.8.13.1~dfsg-3+deb7u1 all Configuration files for Asterisk ii asterisk-core-sounds-en-gsm1.4.22-1 all asterisk PBX sound files - en-us/gsm ii asterisk-modules 1:1.8.13.1~dfsg-3+deb7u1 amd64loadable modules for the Asterisk PBX If the incoming INVITE has the following two multiple bodies then it would not respond to that. It won't even send a Trying. We are using* TCP *only. Content-Type: application/sdp Content-Type: application/ISUP;base=itu-t92+;version=itu-t92+. Is this is a known issue? Are later version of asterisk able to deal with such multi-bodies INVITE? I got to play early media so it needs to make some sense out of first SDP. Best regards, Adnan -- _ -- 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 crash
Is the problem reproducable? /Adnan On Tue, Sep 3, 2013 at 11:17 AM, Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote: Hello List, ** ** In our lab asterisk has crashed due to some unknown reason and it has been restarted by safe_asterisk service. But before crash we can see lots of below log entry (asterisk version 1.8.9.3). ** ** Sep 3 07:55:21] WARNING[16287] res_rtp_asterisk.c: RTP Transmission error of packet to [2002:c117:a683::c117:a683]:20940: Address family not supported by protocol chan_sip.c: Purely numeric hostname, and not a peer--rejecting! ** ** Did someone encounter this problem before? Please let me know. ** ** Regards Rajib -- _ -- 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] WebRTC softphone for Asterisk - any suggestion?
Voxeo/Phono webrtc. /Adnan On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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 -- _ -- 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] Not able to post to list
Hi who is responsible for this mailing list? i am not able to post to it. Br Adnan Sent from my iPhone On 9 okt 2012, at 21:04, Matthew Jordan mjor...@digium.com wrote: On 10/09/2012 02:00 PM, Asterisk Development Team wrote: The Asterisk Development Team has announced the release of libpri 1.4.13. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri The release of libpri 1.4.13 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. (Issue AST-598. Reported by Trey Blancher) * --- Implement handling a multi-channel RESTART request. (Closes issue PRI-93. Reported by Marcin Kowalczyk) * --- Removed MDL/TEI management configuration warning message. (Closes issue PRI-137. Reported by Bart Coninckx) * --- Allow passing compiler flags (CFLAGS, LDFLAGS) (Closes issue PRI-144. Reported by Tzafrir Cohen) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.13 Thank you for your continued support of Asterisk! So as you can tell, this is actually libpri, *not* Asterisk. The script responsible has been sternly reprimanded. Sorry for any confusion! -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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 -- _ -- 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] twenty thousands (20, 000) users, which asterisk and how many servers
the solution lies in kamailio/opensips's despatcher module. Sent from my iPhone On 23 maj 2012, at 20:46, bilal ghayyad bilmar...@yahoo.com wrote: Dear; So it is a hardware issue and not software? I am afraid that asterisk software it self is not able to support 20 000 users and 2000 concurrent calls. About the high availability: is there a method that if the first asterisk server down, then the call will stay connected and failover to second asterisk server? Regards Bilal -- 20.000 users is really a big number, as big as 2000 concurrent calls. As previously stated on this list, it depends... it depends by the type of calls for example. If all media is offloaded from the server letting the phones to reinvite each other, than your server CAN support the call volume. If instead even a tiny portion of the call volume uses service on the pbx, like IVR, music on hold, conferences, queues or even worst, transcoding, then the server is obviously underpowered. From my point of view, servicing 20.000 users with a single piece of hardware is highly risky. It can broke in the middle of the day, leaving all your users without service. I think a better approach will be to have more less powered servers working all together to serving your users. If a day one or two of them broke, you have not to worry because the other will continue to serve your users and nobody notice the little decrease in power. There are a lots of way to achieve the high availability, load sharing, each with its pros and cons. Right now I am building a pbx with high availability and load sharing in mind, for a client who wants to achieve numbers you have just said. Let's see how it works in few months. Leandro 2012/5/23 bilal ghayyad bilmar...@yahoo.com Hi All; I need to use Asterisk for 20 000 users, so which asterisk version to be used? Is there asterisk version that supports 20,000 users on one hardware machine? Can I use one strong hardware server i7 with 64 GB RAM and fast hard desk to handle 20 000 users, and concurrent calls 2000? Or I need multiple servers, how much? If I am going to use multiple servers (until now I do not know how much, and I do not know if the barrier will be the asterisk software or the hardware), then do I have to use special SIP proxy or I have to use load balancer)? In this case, I have to use asterisk Database (so all the servers will read/write from the database)? What about AsteriskNow, can it support? Regards Bilal -- _ -- 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 log format
grep or sgrep Sent from my iPhone On 15 dec 2011, at 18:46, Asterisk Guy arpexpe...@gmail.com wrote: Hi mates! Please, I need to understand how to search for an specific log by date/time on asterisk logs, but can't understand how this works, can you guys please give me an example about how those logs works? Best regards, Asterisk Guy -- _ -- 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] asterisk1.2.1/PRI-E1 outbound call issues
Hello List,I have a following setup:1-Intel Zeon 3.0 Ghz dual Zeon capable board2-Ram 1GB3-OS SLES9 SP24-Asterisk-1.2.15-Wildcard TE110P(Using as E1) 6-Wildcard TDM03BPRI/E1 is up and running perfectly inbound /outbound calls goes perfectly in start but after sometime almost all outbound calls disconnected/hangup automatically for example user is dialout and he/she is talking then after sometime call disconnected inbound calls are also facing same issue but rate is very low sometimes while outbound calls becomes nightmare any idea here is my config:zaptel.conf:span=1,0,0,ccs,hdb3,crc4,yellowbchan=1-15,17-31dchan=16#dchan=31fxsks=33-35loadzone = usdefaultzone=us zapata.conf:[channels];calleridasreceived=yesusecallerid=yeshidecallerid=no;callwaitingcallerid=yes;threewaycalling=yes transfer=yescallwaiting=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesechotraining=800relaxdtmf=yes;faxdetect=incomingimmediate=no;Callgroup=1;Pickupgroup=1; ;context=from-pstn;switchtype=nationalsignalling=pri_cpe;faxdetect=incomingusecallerid=yescidsignalling=bellcidstart=ringpridialplan=nationalprilocaldialplan=nationalnationalprefix=0 localprefix=021busydetect=yesechocancel=yescallerid=yesechocancelwhenbridged=yesechotraining=800group=1channel=1-15,17-31;Callgroup=1;Pickupgroup=1context=incoming-analog group=2signalling=fxs_kschannel=33-35 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video Conferencing
On 1/1/06, Nir Simionovich [EMAIL PROTECTED] wrote: Well, the documentation states that Video Conferencing is possible. I'vetried working with EyeBeam, which yielded nice Results, but anything beyondthat - I can't comment.Nir Scan you share your experience with us i.e. what asterisk version what camera/webcam you r using.-Original Message- From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ] On Behalf Of DakotaSent: Sunday, January 01, 2006 10:21 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Video ConferencingCan the asterisk system support video conferencing? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
On 8/6/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Peter Svensson wrote: On Sat, 6 Aug 2005, Robert Goodyear wrote:Can you educate us all on the appropriate circumstances in which touse 'r'? Some devices (voip phones, softphones) do not generate in band progress information when ringing. You will quickly find out if a particular end device requires the 'r' option or not. You almost never want it enabled on a trunk line, only for terminal devices. Almost nothing generates inband ringing.That has nothing to do with r.--Eric Wieling * BTEL Consulting * 504-210-3699 x2120r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insertthis by default into all your dial statements as you are killing callprogress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically whereit is appropriate to do so. r makes it go the next step andadditionally generate ring tones where it is probably not appropriate to do so.That's great but i have few things to asking!We have 4 servers = User1 move from server1 to server2 ,he registers on server2. Dials an extension let's say 100 ,so all calls for User1 route on that extension.Remember call comes from any of the 4 servers. I implements that sort of functionality in different way but really want that sort of dial plan is that possible or i am asking a dumb question ___Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk wiht LDAP
I am trying to configuring/running Asterisk::LDAP perl module getting from http://projects.alkaloid.net/ but no luck i have successfully installed this module but when i include its scheme file which is asterisk.scheme in the LDAP include list and try to start the LDAP Server service its gives the following error: /etc/init.d/ldap start Starting ldap-server/etc/openldap/schema/asterisk.schema: line 181: Unexpected token before 1.3.6.1.4.1.1466.115.121.1.36 EQUALITY numericStringMatch ) ObjectClassDescription = ( whsp numericoid whsp ; ObjectClass identifier [ NAME qdescrs ] [ DESC qdstring ] [ OBSOLETE whsp ] [ SUP oids ]; Superior ObjectClasses [ ( ABSTRACT / STRUCTURAL / AUXILIARY ) whsp ] ; default structural [ MUST oids ] ; AttributeTypes [ MAY oids ]; AttributeTypes whsp ) startproc: exit status of parent of /usr/lib/openldap/slapd: 1 failed Its include path is : /etc/openldap/schema/asterisk.schema But when i comment this line from LDAP Server started successfully can anyone trying this app kindly helping me out ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk wiht LDAP
I am trying to configuring/running Asterisk::LDAP perl module getting from http://projects.alkaloid.net/ but no luck i have successfully installed this module but when i include its scheme file which is asterisk.scheme in the LDAP include list and try to start the LDAP Server service its gives the following error: /etc/init.d/ldap start Starting ldap-server/etc/openldap/schema/asterisk.schema: line 181: Unexpected token before 1.3.6.1.4.1.1466.115.121.1.36 EQUALITY numericStringMatch ) ObjectClassDescription = ( whsp numericoid whsp ; ObjectClass identifier [ NAME qdescrs ] [ DESC qdstring ] [ OBSOLETE whsp ] [ SUP oids ]; Superior ObjectClasses [ ( ABSTRACT / STRUCTURAL / AUXILIARY ) whsp ] ; default structural [ MUST oids ] ; AttributeTypes [ MAY oids ]; AttributeTypes whsp ) startproc: exit status of parent of /usr/lib/openldap/slapd: 1 failed Its include path is : /etc/openldap/schema/asterisk.schema But when i comment this line from LDAP Server started successfully can anyone trying this app kindly helping me out ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Single extension/user registers across multiple asterisk servers
Hello , I have a question which i am not clear that whether it is possible or not so i want some help to clearify Sorry for very long mail: we have eight asterisk servers across different cities connected through IAX intenet connection is DSL broadband so for sake simplicity and easiness for eight servers i assigned particular/fixed extensions dial patterns for example 1XXX for Server 1,2XXX for server 2 and so on upto eight servers this setup works fine now one user let say register on server 1 move another location for example in server 2 domain what i want to do when user1 moves any location upto 7 other locations its user/extension remain same i.e. SIP/1001 and when this user connected it becomes local user but its identitity not change . The reason for doing that if user1 moves to any other location it registers to its own server i.e. server1 and it is physically connected in the domain of server2 so user1 dials the 2XXX extensions i.e. on server2 the call first goes to server1 which resides in another city/location then server1 routes the call to server2 logically this is right and exact way but physically it is very funny is this possible means dial plan sharing every single user registered on multiple locations automatically everytime he/she connected to that location so in this way he/she will be the local user for that location and saves lot of precious bandwidth . Sorry for very long question but right now i am stuck i think this is possible but how i am looking wiki alot DUNDI,ENUM etc but fails to decided whether it is possible or not. Sorry for bothering Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CRM integration (was RE: CallerID)
On 5/31/05, Anton Krall [EMAIL PROTECTED] wrote: I am doing some testing using FOP (Flask Operator Panel) and so far, its going great! Been able to do callerid and also open a SugarCRM screen. All without having to install anything on the computer, just open a FOP browser screen and that's it! More later when I debug some ideas. Anton can you show it how you integrate FOP with SugarCRM i also try it but not successed but really need it because SugarCRM is far sophasticated and robust than anyother CRM package sorry for asking u because my main problem is that i am not a php programmer. Thanks in Advance. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Adam Goryachev |Sent: Lunes, 30 de Mayo de 2005 09:18 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] CRM integration (was RE: CallerID) | |On Sat, 2005-05-28 at 19:19 +0100, Tom Fanning wrote: | snip | | The guy mentioned Java from within the browser. I believe that I am | right in saying that a Java applet should very well be able |to listen | for tcp connections as well as udp datagrams. Try this primer: | |http://homepages.uel.ac.uk/2795l/pages/javaapps.htm#Class%20ServerSock | et%20( | TCP%20Server%20Connections) | |Yep, thanks for replying for me... | |So, has anyone got the time + motivation to do something??? I |wish I did :( | |Regards, |Adam | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk integration with Quintum Tenor AXT800!
Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio? i have a scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for intranet no PSTN at all just only IPphones connected through ehternet port and analog phones connected on FXS port.Is it's neccassary to connect with PSTN i don't want PSTN DIALING i'll just internal dialing.the scenerio is ipphones connected through ethernet while analog phones directly connected through FXS port is that possible i integrate Tenor AXT 800 in such a way that i describe above or may be i am asking a blind n dumb question Thw model number of voip gateway is Quintum Tenor AXT 800 with 8FXO,8FXS and 10/100Mbs LAN port. Kindly comments on that whether is that possible or not or what is the best way to utilize the power of Tenor gateway,practical experience working implementationc etc. Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quintum Tenor AXT800!
Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio i have scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for interanet no PSTN at all just only IPphones and analog phones connected on FXS port.Is it's neccassary to cannect with PSTN i don't want i'll just want to use my internal/company premesis. ipphones connected through ethernet while analog phones directly connected through FXS port is that possible i integrate Tenor AXT 800 in such a way that i describe or may be i am asking a blind n dumb question Thw model number of voip gateway is Quintum Tenor AXT 800 with 8FXO,8FXS and 10/100Mbs LAN port. Kindly comments on that whether is that possible or not or what is the best way to utilize the power of Tenor gateway,practical experience working implementationc etc. Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX-IAX Trunking not works
no man iax2 trunking not working i don't know why its really odd iax2 trunk debug command shows IAX2 Trunk Debug Requested Beginning trunk processing Ending trunk processing with 0 peers and 0 calls processed wat's that means how can i enable trunking on one ser iax2 show channels command shows: asteriskser1*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format IAX2/[EMAIL PROTECTED] 192.168.0.151test2 3/3 00121/00108 00020ms 0006ms 0056ms gsm IAX2/[EMAIL PROTECTED] 192.168.0.151test2 6/8 6/3 00013ms 0001ms 0049ms gsm on another server shows test2*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format IAX2/[EMAIL PROTECTED]/2192.168.0.77 adnan 2/20687 00026/00023 [Native Bridged to ID=4] IAX2/192.168.0.51:45 192.168.0.51 test2 4/4 00021/00025 [Native Bridged to ID=2] IAX2/[EMAIL PROTECTED]/5 192.168.0.79 iphone 5/25617 00025/00024 [Native Bridged to ID=6] IAX2/192.168.0.51:45 192.168.0.51 test2 6/3 00021/00026 [Native Bridged to ID=5] 4 active IAX channel(s) is something going wrong plz i am very keen to solve this as soon as possible plz kindly enlighten on this issue. IAX2 Trunk Debug Requested Beginning trunk processing Ending trunk processing with 1 peers and 3 calls processed If you want to free up more bandwidth add echocancel=no to your iax.conf Gary Lawrence -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Clark Sent: Monday, May 23, 2005 10:01 AM To: Adnan Ahmed; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX-IAX Trunking not works Adnan Ahmed wrote: Hello , I want some tips guidance i am sure this topic discuss alot in list,i try my best to solve it by myself try googling looking wiki everywhere but no luck question is iax-iax trunking not working setting,trying each n every option server2 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [saim] username=saim secret=saim type=friend host=dynamic context=from-sip disallow=all allow=gsm [noman] username=saim secret=noman type=friend host=dynamic context=from-sip disallow=all allow=gsm [asteriskser1] type=friend ;auth=md5 ;secret=qwerty context=local ;host=dynamic defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no server1 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [user1] username=user1 secret=user1 type=friend host=dynamic context=from-sip disallow=all allow=gsm [user2] username=user2 secret=user2 type=friend host=dynamic context=from-sip disallow=all allow=gsm [test2] type=friend context=local defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no I am using Kiax soft phone on both servers using codec GSM asterisk latest stable version OS SLES9 ,any help is highly appreciated i had look almost every place in wiki regarding iax trunking but all in vein. Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please don't use reply when you are starting a new thread. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX-IAX Trunking not works
Hello , I want some tips guidance i am sure this topic discuss alot in list,i try my best to solve it by myself try googling looking wiki everywhere but no luck question is iax-iax trunking not working setting,trying each n every option server2 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [saim] username=saim secret=saim type=friend host=dynamic context=from-sip disallow=all allow=gsm [noman] username=saim secret=noman type=friend host=dynamic context=from-sip disallow=all allow=gsm [asteriskser1] type=friend ;auth=md5 ;secret=qwerty context=local ;host=dynamic defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no server1 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [user1] username=user1 secret=user1 type=friend host=dynamic context=from-sip disallow=all allow=gsm [user2] username=user2 secret=user2 type=friend host=dynamic context=from-sip disallow=all allow=gsm [test2] type=friend context=local defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no I am using Kiax soft phone on both servers using codec GSM asterisk latest stable version OS SLES9 ,any help is highly appreciated i had look almost every place in wiki regarding iax trunking but all in vein. Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax trunking not works!
hello, iax trunking not working we actually testing dial 500(Digium) two or three calls simultaneously but bandwidth graph shows 95 to 100kbps not match the results shows on wiki iax bandwidth pages i enable trunk=yes in iax.conf is there any tweaking or optimization because i desperately need some solution for this Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax trunking not works!
sorry for incomplete info i am using GSM codec while softphones are kiax i try three calls simultaneously each channel took 35kbps it is really odd three channel consumes almost 95 to 100kbps sounds preety odd i am in deep trouble now because soon we implement several servers on remote locatons connected to each other through IAX and if trunking not works its became a nightmare anyone suggest some other codecs except GSM ,i'll be very thankful. Thank You. Adnan Ahmed. On 5/13/05, Jay Milk [EMAIL PROTECTED] wrote: What codec are you using? -Original Message- From: Adnan Ahmed [mailto:[EMAIL PROTECTED] Sent: Friday, May 13, 2005 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] iax trunking not works! hello, iax trunking not working we actually testing dial 500(Digium) two or three calls simultaneously but bandwidth graph shows 95 to 100kbps not match the results shows on wiki iax bandwidth pages i enable trunk=yes in iax.conf is there any tweaking or optimization because i desperately need some solution for this Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with Suse Linux Enterprise !
Hello, i am running suse linux enterprise edition of kernel version 2.6.5-7.97-smp, i have latest stable asterisk zaptel asterisk stuff compile fines i have TDM400P card with 1FXS and 3FXO modules, every time i probe with modprobe and issue ztcfg -vv commandit shows the following errors: also issue modprobe wcfxs but no luck asterisk2:/lib # modprobe zaptel asterisk2:/lib # modprobe wct4xxp asterisk2:/lib # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) i am also sets udev configuration files udev.rules and permissions.udev as describe on wiki am i doing something wrong. please i have want some quick tips suggestions guidelines. zaptel.conf: fxoks=1 fxsks=2-4 loadzone = us defaultzone=us Thanks in advance please helping me out. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] want just few words from the list about SLES!
Hello, Sorry for bothering again i asked it early but noreply may be swap now ask it again hopefuly this time not vein my question is anyone try installing/running on Suse Linux Enterprise Server v9 ,may be vey helpful for me i try installing suse 9.2 professional but not successful not try on SLES but not sure about it . Sorry again for bothering but kindly requested ot. Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with SLES!
Hello, sorry again i send a mail early which i can't receive so i send it again i am trying to install asterisk om suse linuex enterprise server but can;t make it also try udev settings but not working can anyone successfully installed asterisk on SLES helps a lot. Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk's on Suse Linux Enterprise Server(SLESv9)
hello, can anyone installing/configuring asterisk's on SLES9 if someone can share his/her views experiences . Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astguiclient error!
Hello, can anyone using astgui client i have a problem in installation phase everytime i try to create database from MySQL_AST_CREATE_tables.sql it gives error in phone table ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'DBY_server VARCHAR(15), DBY_database VARCHAR(15) default 'asterisk', DBY_user VA' at line 62 i also try manually to create this table but no luck am i missing something ? Thanks In Advance ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting Multiple Asterisk Servers!
Hello, We 'll setup asterisk servers on several remote locations atleast 6-7 different locations these are connected to each other through DXX(Digital Cross Connect) ,on larger locations we use PRI/E1 and small locations we use TDM400 may be one or two but lot of IP phones(soft/hard phones),basically we are currently in planning phase to which one is the best for implementing this setup either we use IAX to connect asterisk servers together implement DUNDi using IAX/SIP using SER(SIP Express Router) which is the best can anyone implement this type of setup before may be very helpful for guiding the best scalable robust setup because in future may be this setup expands so we also take a look at it any suggestions,tips,guidelines,weblinks may be very helpful. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting Multiple Asterisk Servers!
William, Thanks friend for your reply we have following setup: ipphones--routerasterisk server with channel bank loaded with Quad T1 card, also on channel bank several analog phones connected we deploy these setup at our headoffice and other branch offices situated in different cities we use TDM400 cards. Basiacally what we want to do we made our organization own internal network nationwide provides ivr,voicemail,call transfering,call waiting.confrencing etc. possible solutions are: ---we connected these servers through IAX ---we Implement DUNDi using IAX/SIP ---using asterisk+SER but not sure how? which one will be a best and scalable solution any suggestios,tips,guidelines etc Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fxo card not workin in susev9.2!
Hello, I am new in linux and also suse i have a fxo card but its not working the errors are: Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Notice: Configuration file is /etc/zaptel.conf line 143: Unable to open master device '/dev/zap/ctl' Mar 13 21:35:13 linux kernel: zaptel: unsupported module, tainting kernel. Mar 13 21:35:13 linux kernel: Zapata Telephony Interface Registered on major 196Mar 13 21:35:20 linux kernel: wcfxo: unsupported module, tainting kernel. also try udev as mentioned on wiki but not working can anybody really configuring asterisk on suse9.2 with or without hardware kindly help me . Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] i am missing something!
thanks for replying but no change at all any other tips,suggestions thanks in advance On Wed, 9 Mar 2005 01:44:41 -0600, Jay Milk [EMAIL PROTECTED] wrote: You'll need canreinvite=no to each sip section in sip.conf, if you want * to stay in the loop. -Original Message- From: Adnan Ahmed [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 09, 2005 1:14 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] i am missing something! Hello ppl, At initial level i configure asterisk woth only soft phones ,in which one at windows machine and other is linux i am using windows messenger and linphone respectively both phones registered with asterisk respectively problem is that they bypass asterisk on call when i send request from linphone to messenger request shown on messenger but on asterisk console nothing to and also if i send request from messenger to linphone it doesn't recognized at all my config are: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] i am missing something!
Hello ppl, At initial level i configure asterisk woth only soft phones ,in which one at windows machine and other is linux i am using windows messenger and linphone respectively both phones registered with asterisk respectively problem is that they bypass asterisk on call when i send request from linphone to messenger request shown on messenger but on asterisk console nothing to and also if i send request from messenger to linphone it doesn't recognized at all my config are: extensions.conf: [general] static=yes writeprotect=no [sip] exten = 101,1,Dial(SIP/101,20) msn exten = 922,2,Dial(SIP/102,20) -linphone sip.conf: [general] context=sip port=5060 bindaddr=192.168.0.50 (asterisk server ip) maxexpirey=3600 defaultexpirey=120 disallow=all allow=ulaw allow=alaw allow=gsm relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 ;useragent=Asterisk PBX ;nat=no [911] username=101 type=friend callerid=101 context=sip qualify=no host=dynamic nat=no canreinvite=yes disallow=all allow=ulaw allow=alaw ;allow=gsm defaultip=192.168.0.60 [912] username=102 type=friend host=dynamic dtmfmode=inband context=sip disallow=all allow=alaw allow=ulaw ;allow=gsm nat=no defaultip=192.168.0.51 canreinvite=yes what i want when asterisk registers it can only make calls otherwise refuse it . Don't bother with my question. Thank You. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] video confrencing
Hello *'s, I have a question regarding asterisk in asterisk is video confrencing is possible like meetme i am out of touch quite a long time so don't bother with my question if video confrencing is possible what kind of hardware required i already working on softphones setup with asterisk including video support . Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wrong CVS version ?
you are compiling in wrong sequence first zaptel then asterisk and after that asterisk-addons . hope this helps ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP with Asterix
Dana Olson wrote: -- voip-info.org On Mon, 31 Jan 2005 20:21:51 -, Richard Dutton [EMAIL PROTECTED] wrote: Hi Guys, I know no doubt this has been covered on the list a zillion time before, but can anyone point me to some good resources on using Asterix as a VoIP gateway? I would like to get two TDM400P cards in two machines attached to separate adsl connections (in two different physical locations). I'd then like to be able to plug POTS telephones into each of them (I understand I will need FXS interfaces in both) and be able to make calls between them. Is this hard to set up? Are there any guides that you guys would recommend? I would then like to attach a POTS line to one side and be able to dial into my asterix, and out to the other side across the adsl. What codecs do you guys use for VoIP and what sort of quality can you get over regular 512mb adsl? Any help would be most appreciated, Asterix looks like an excellent system and I can't wait to get started with it! Cheers Rich -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.2 - Release Date: 28/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users check-it-out http://www.voip-info.org/tiki-index.php?page=Asterisk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk
just finish it if anyone like mysql go for it or someone love postgresql its ok but don't ruin the purpose of this list keep out these kind of mess sorry areski for that and thanks for your great work ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC
Bilal Ghayad wrote: Dear Sebastian; Thanks a lot for your kindly advise to use ASTCC. But can u advise me the link for ASTCC to download it and wether it is open source (to download the source and work on it? Regards Bilal _ check it out http://www.voip-info.org/wiki-ASTCC regards __ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with astcc
hello *'s,
Astcc not workin what is correct format for defining
1-database
2-brands
3-trunks
4-routes
i define all these things but not workin may be i define in wrong
format.I have FXO card installed.can anyone implement it and also my
sip phone generates very loud noise wat is that i tried several
settings but not hear any voice just noise.
sip.conf
[general]
context=from-sip
port=5060
bindaddr=192.168.10.186
disallow=al
allow=gsm
musicclass=default
relaxdtmf=yes
register = sipuser:[EMAIL PROTECTED]
[101]
username=101
type=friend
secret=secret
host=dynamic
defaultip=192.168.10.176 ;ip addr of sip phone
disallow=all
allow=gsm
nat=yes
qualify =1000
[adnan.007]
username=adnan.007
type=peer
secret=secret
host=iptel.org
disallow=all
allow=gsm
nat=yes
qualify=1000
extensions.conf
[general]
static=yes
writeprotect=no
#include = /var/lib/astcc/astcc-exten.conf
[incoming]
exten = _N.,1,Dial(Zap/1)
exten = _N.,2,DeadAGI(astcc.agi)
exten = _N.,3,Hangup
[from-sip]
adnan.007=101
exten = _N.,1,DeadAGI(astcc.agi,${ACOUNTCODE},${EXTEN})
exten = _N.,2,Hangup
Thanks In Advance.
Adnan Ahmed.
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[Asterisk-Users] is wiki drunk
is there any problem with wiki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RealTime Drivers Connectivity Error
Hello *'s, i am using Realtime Sip drivers but its not working here is my configs: extconfig.conf [settings] ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; sipfriends = mysql,asterisk,sip_friends res_mysql.conf [general] dbhost = localhost.localdoamin/127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = 123456 dbport = 3306 dbsock = /var/lib/mysql/mysql.sock error detail: Dec 31 01:20:49 ERROR[4298]: res_config_mysql.c:617 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on localhost.localdomain/127.0.0.1. Check debug for more info. == Registered application 'UserEvent' [app_verbose.so]Segmentation fault (core dumped) i change dbhost parameter several times like(localhost,192.168.10.193 etc) but can't works I am using latest CVS-Head kindly pointout my mistakes. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation Fault (core dumped)
i am facing unusual and wiered error in asterisk using Realtime MYSQL driver . Asterisk runs well and smoothly with absoulutely no error or warning but everytime i power-on my sip-phone ,booting, initializes and then asterisk suddenly quit with the error. _*Segmentation Fault (core dumped)*_ i see in /var/log/messages,/var/log/asterisk/messages but all is clear no sign of any error message or warning, what does its mean its my configs problem or something wrong with asterisk i use Latest CVS. Can i use Realtime odbc instead of Mysql . extconfig.conf [settings] ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; sipfriends = mysql,asterisk,sip_friends res_mysql.conf [general] dbhost = localhost.localdoamin/127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = 123456 dbport = 3306 dbsock = /var/lib/mysql/mysql.sock ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help regarding ASTCC
Hello *'s, I just installed and running successfully _*Mark's*_ *ASTCC* calling card application.But there is no documentation out there about usage of this application can anyone using this application. There are lot of options like Brands,Cards,Trunks etc its web layout is preety good but no help/documentation. Thanks Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MYSQL_FRIENDS
Hello *'s, Hi, I've just tried to enable MYSQL Friends in CVS HEAD. But i cannot find this option.On wiki i found this. To enable this, you need to edit the Makefile in the channels directory of your source tree and enable MYSQL_FRIENDS. This enables database definition of both IAX2 and SIP friends. Make sure you have the MySQL development kit (libraries) installed before compilation.But where is MYSQL_FRIENDS option.I can't find it.I used Latest CVS. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk at large
Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i do this or may be i changed my plans kindly guides me. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] calling card application
can anyone using/integrating modified-prepaid-application avaiable on wiki . if anyone kindly guided me. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] modified prepaid application
how can we integrate the modified-prepaid application with asterisk because when i compile the app_prepaid it gives bunch of errors. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk With PostgreSQL
Hi *'s, Back Again I want to use PostgreSQL instead of MySQL, basically i want to create an application (calling card),firstly i am not be able to connecting postgres to my asterisks i made some configuration in odbc.ini,odbcinst.ini cdr_pgsql etc but no luck asterisk doesn't recognize it unknown host name error displayed my asterisk's server name is asterisk also my database name is asterisk ,searches wiki alot but almost all info about MySQL i have facing difficulties to connect PostgreSQL from asterisk what files i'll change to properly setup PostgreSQL with Asterisk any help is highly appreciated. This is my second post on this issue no responce on first one so plz take a while . Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Workimg On PostgrSQL
Hi *'s, Back Again I want to use PostgreSQL instead of MySQL basically i want to create an application (calling card),what is the procedure i mean in which files i saw several config files and change it slightly but not sure about it i search wiki alot but on wiki almost all info about MySQL actually i have facing difficulties to connect PostgreSQL from asterisk what files i'll change to properly setup PostgreSQL with Asterisk any help is highly appreciated. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk's Empty Folder
Hello *'s, I have recently installed CentOS v3.3 and i have latest stable Asterisk's source code ,i compiles it shows no error but when i am looking for sip.conf,zapata.conf ,i am amazed the /etc/asterisk folder was empty i am compling several times but no luck what's the problem i compiled in the order of zaptel,libpri,asterisk. I send some traces of my asterisk's compilation kindly pointout my mistakes. o synths.o synths.c synths.c:172: warning: no previous prototype for `synths_' synths.c: In function `synths_': synths.c:401: warning: implicit declaration of function `irc2pc_' synths.c:402: warning: implicit declaration of function `bsynz_' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I ../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-12/08/04-14:03:10\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\ when i compile asterisk these errors are aoming several times in several files and at the end + Asterisk Installation Complete ---+ +YOU MUST READ THE SECURITY DOCUMENT+ + Asterisk has successfully been installed. + + If you would like to install the sample + + configuration files (overwriting any + + existing config files), run: + + make samples+ kindly pointout what's wrong i am doing bocz i spend almost a day or above but all in vein. Thanks in Advance Adnan Ahmed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk's Empty Folder
Jim Radford wrote: You need to do a: make install and then make samples to install sample conf files. Jim On Wed, 8 Dec 2004, Adnan Ahmed wrote: Hello *'s, I have recently installed CentOS v3.3 and i have latest stable Asterisk's source code ,i compiles it shows no error but when i am looking for sip.conf,zapata.conf ,i am amazed the /etc/asterisk folder was empty i am compling several times but no luck what's the problem i compiled in the order of zaptel,libpri,asterisk. I send some traces of my asterisk's compilation kindly pointout my mistakes. o synths.o synths.c synths.c:172: warning: no previous prototype for `synths_' synths.c: In function `synths_': synths.c:401: warning: implicit declaration of function `irc2pc_' synths.c:402: warning: implicit declaration of function `bsynz_' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I ../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-12/08/04-14:03:10\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\ when i compile asterisk these errors are aoming several times in several files and at the end + Asterisk Installation Complete ---+ +YOU MUST READ THE SECURITY DOCUMENT+ + Asterisk has successfully been installed. + + If you would like to install the sample + + configuration files (overwriting any + + existing config files), run: + + make samples+ kindly pointout what's wrong i am doing bocz i spend almost a day or above but all in vein. Thanks in Advance Adnan Ahmed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks *'s its working thankyou very much all of you ppls. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Contact me Asap!
Hello Khurram, This is adnan from EBS kindly contact me as soon as possible i'll contact you on your number but its almost busy every time. Other *'s users kindly forgive me because i have no option right now. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't Register!
Hello *'s, Every time I started my asterisk error shows: chan_sip.c:Got 200 ok onRegister that isn't a register. Failed to authenticate in Register to sip:[EMAIL PROTECTED];tag=asfeaa71f Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 107 (Request) Registration for [EMAIL PROTECTED] timed out ,trying again. __sip__xmit:sip_xmit of 0x810894 (len 436) to 195.37.77.99 returnes -l:Invalid Argument I calls locally from my sip phone easily and from local phone to sip phone but i can't calls long distance every time itries above error shows on my screen am i missing something in my sip.conf file. sip.conf [general] port=5060 bindaddr=192.168.10.189 context=sip disallow=all allow=gsm nat=1 [101] username=101 type=friend host=dynamic secret=xyz context=from-sip callerid=101 dtnfmode=rfc2833 canreinvite=no qualify=1000 [iptel] username=adnan.007 type=friend secret=123 host=dynamic fromdomain=iptel.org qualify=1000 nat=1 context=outgoing May be I am doing something wrong kindly pointout my mistakes and clear my concepts. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Problem!
hi,
I am not registered my SIP Phone with Asterisk i spend almost one day
but find no luck.I know very well this is not kind a problem discussed
in this group but i try my best and all in vein so finally i am here
hoping you ppl helping me out.I discussed this problem in
asterisk's-users group and adding feedback from asterisk-users group my
configs are
sip.conf
[general]
port=5060
bindaddr=192.168.10.193
allow=all
[101]
username=101
type=friend
secret=12345678
host=192.168.10.193
context=from-sip
callerid=101101
defaultip=192.168.10.176
extensions.conf
[globals]
101=SIP/101
[incoming]
exten = s,1,Dial(Zap/1,20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${announce})
exten = s-NOANSWER,2,Goto(incoming,s,1)
exten = s,3,NoOp,$(CALLERID)
include = outgoing
include = from-sip
callerid=yes
[outgoing]
exten = _NXX,1,Dial/Zap/4/${EXTEN:0}
exten = _0N,1,Dial,Zap/4/${EXTEN:0}
exten = _0NX,1,Dial,Zap/4/${EXTEN:0}
exten = _0NXX,1,Dial,Zap/4/${EXTEN:0}
exten = 101,1,Dial(101,20)
include = from-sip
include = incoming
[sip]
exten = 101,1,Dial(${101,20})
exten = 101,2,VoicemailMain
exten = 101,3,Hangup
include = outgoing
include = from-sip
here are the console output : :-X ).
*cli --Starting simple switch on 'Zap/1-1'
Executing Dial( ,) in
new stack
Called 101
Got SIP Responce 482 Loop Detected back from 192.168.10.193
No one is available to answer qt this time
Executing VoiceMailMain( ,) in new stack
Playing 'vm-login' (language 'en' )
Username not entered
Executing Hangup( ,) in new stack
Spawn Extension (outgoing , 101, 3) exited non-zero on 'Zap/1-1'
Hangup 'Zap/1-1'
*clisip show registry
Host Username
Refresh State
*clisip show users
Username Secret Authen
Def.Context A/C
101 12345678md5,plaintext
sipNo
*clisip show peers
Name/UsernameHost Mask
Port Status
101/101192.168.10.195255.255.255.255
5060Unmonitored
*clisip show channels
PeerUser/ANRCall IDSeq
(Tx/Rx) LagJitterBuffer
0 active SIP channel(s)
Kindly pointout my mistakes/errors and helping me out.
Any Help Is Highly Appreciated.
Thanks in Advance.
Adnan Ahmed.
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[Asterisk-Users] Re:SIP Problem
I am very thankful to you people for helping me as much i imagine but i still
need your help, problem is that i am not be able to dial from my analog phone
conected to fxs card to my sip phone i change my configs but still no result.
sip.conf
[general]
port=5060
bindaddr=192.168.10.193
allow=all
[101]
username=101
type=friend
secret=12345678
host=dynamic
context=from-sip
callerid=101101
defaultip=192.168.10.176
extensions.conf
[globals]
101=SIP/101
[incoming]
exten = s,1,Dial(Zap/1,20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${announce})
exten = s-NOANSWER,2,Goto(incoming,s,1)
exten = s,3,NoOp,$(CALLERID)
include = outgoing
include = from-sip
callerid=yes
[outgoing]
exten = _NXX,1,Dial/Zap/4/${EXTEN:0}
exten = _0N,1,Dial,Zap/4/${EXTEN:0}
exten = _0NX,1,Dial,Zap/4/${EXTEN:0}
exten = _0NXX,1,Dial,Zap/4/${EXTEN:0}
exten = 101,1,Dial(101,20)
include = from-sip
include = incoming
[sip]
exten = 101,1,Dial(${101,20})
exten = 101,2,VoicemailMain
exten = 101,3,Hangup
include = outgoing
include = from-sip
here are the console output : :-X ).
*cli --Starting simple switch on 'Zap/1-1'
Executing Dial( ,) in
new stack
Called 101
chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqnp 102 (Request)
No one is available to answer qt this time
Executing VoiceMailMain( ,) in new stack
Playing 'vm-login' (language 'en' )
chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqnp 102 (Request)
Username not entered
Executing Hangup( ,) in new stack
Spawn Extension (outgoing , 101, 3) exited non-zero on 'Zap/1-1'
Hangup 'Zap/1-1'
Thanks in Advance.
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Re: [Asterisk-Users] Newbie Question
Leo Salas wrote:
I am just learing some Linux and have been able to setup Asterisk
samples and channels fxo card on ch.1 and fxs on ch 4.
I have an Internet Polycom phone to use to test to/from internet and 1
analouge phone connected to port 4 of Digium TDM-400 with appropriate
cards installed to dial out on. I wish to dial to the outside via
PTSN line. I am lost on the instructions. Can anyone help with
Extensions.conf and sap.conf. 3 extensions are needed.
Thanks for help.
Leo
I am using same setup running smoothly i am sending you my configs file
hope you enjoy it currently i am dialing my sip phone to pstn and sip to
dial anywhere in the world and also via pstn i am dialing my sip as well
as locally fairly comfortably.
you change as per your requirement don't just copy try to understand the
concept,logic behind that and you will be a happy man.
extensions.conf
[general]
static=yes
writeprotect=no
[outgoing]
exten = _XX,1,Dial(Zap/4/${EXTEN})
exten = _111XX,1,Dial(Zap/4/${EXTEN})
exten = _1X,Dial(IAX Account)
exten = _1X,Dial(SIP Account)
exten = 103,1,Dial(Zap/1)
exten = 123,1,VoiceMailMain
exten = 101,1,Dial(SIP/101,20)
[from-sip]
include = outgoing
sip.conf
[general]
port=5060
bindaddr=192.168.10.193
context=sip
disallow=all
allow=gsm
nat=1
[101]
username=101
type=friend
host=dynamic
secret=1234
context=from-sip
callerid=101
dtmfmode=rfc2833
canreinvite=no
;disallow=all
;allow=gsm
qualify=1000
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Re: [Asterisk-Users] I Am Missing Something Somewhere Somehow!
Mike Dent wrote: Dont get caught by the same thing which had me ripping my hair out! I had installed Fedora core 2 on a box and forgot that it had installed iptables firewall! Type iptables -L and see if there are any rules? iptables -F will flush them for the time being, then try again. It worked for me, wow how silly I felt! Mike I am using Debian it's not working for me any other thaughts,tips suggestions because now i am very exhausted with this error i am looking almost everyplace wiki google but no luck kindly helping me out. On Sun, 21 Nov 2004 23:23:49 +0500, Adnan Ahmed [EMAIL PROTECTED] wrote: el Flynn wrote: Adnan Ahmed wrote: hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck my configs are. snip *clisip show peers Name/UsernameHost Mask Port Status 101/101192.168.10.195255.255.255.255 5060Unmonitored your sip show peers command shows that the phone is indeed connected to your Asterisk server. If you are having problems doing stuff with it, may I suggest you changing your dialplan to the following just to test things out: [sip] exten = 1,1,VoicemailMain exten = 1,2,Hangup then restart asterisk and dial 1 from your SIP phone. If you can hear the voicemail application prompts then you're okay. flynn *It's not working still in silence mode don't show anything don't do anything any other suggestions,tips ,configs .* Thanks In Advance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I Am Missing Something Somewhere Somehow!
Mike Dent wrote: Did you try the iptables -L as I suggested though? It's probably still present in Debian. Mike On Mon, 22 Nov 2004 02:47:53 +0500, Adnan Ahmed [EMAIL PROTECTED] wrote: Mike Dent wrote: Dont get caught by the same thing which had me ripping my hair out! I had installed Fedora core 2 on a box and forgot that it had installed iptables firewall! Type iptables -L and see if there are any rules? iptables -F will flush them for the time being, then try again. It worked for me, wow how silly I felt! Mike I am using Debian it's not working for me any other thaughts,tips suggestions because now i am very exhausted with this error i am looking almost everyplace wiki google but no luck kindly helping me out. *yes mike i am trying it both of your commands but still no luck.* ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I Am Missing Something Somewhere Somehow!
hi,
I am not registered my SIP Phone with Asterisk i spend almost one day
but find no luck my configs are.
sip.conf
[general]
port=5060
bindaddr=192.168.10.195
disallow=all
allow=alaw
allow=ulaw
[101]
username=101
type=friend
secret=1234
host=192.168.10.195
context=sip
callerid=101101
defaultip=192.168.10.176
extensions.conf
[globals]
[incoming]
exten = s,1,Dial(Zap/1)
[outgoing]
exten = _NXX,1,Dial/Zap/4/${EXTEN:0}
exten = _0N,1,Dial,Zap/4/${EXTEN:0}
exten = _0NX,1,Dial,Zap/4/${EXTEN:0}
exten = _0NXX,1,Dial,Zap/4/${EXTEN:0}
exten = 101,1,Dial,Zap/4(SIP/101)
[sip]
exten = 101,1,Dial(SIP/101,20)
here are the console output : show no errors but also not working
(running Asterisk in quite mode :-X ).
*clisip show registry
Host Username
Refresh State
*clisip show users
Username Secret Authen
Def.Context A/C
101 12345678md5,plaintext
sipNo
*clisip show peers
Name/UsernameHost Mask
Port Status
101/101192.168.10.195255.255.255.255
5060Unmonitored
*clisip show channels
PeerUser/ANRCall IDSeq
(Tx/Rx) LagJitterBuffer
0 active SIP channel(s)
kindly pointout my mistakes/errors and helping me out.
I am searching wiki,google but no luck i am tried several configs but
all in vein please please helping me out :-( .
Thanks In Advance .
Adnan Ahmed.
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Re: [Asterisk-Users] I Am Missing Something Somewhere Somehow!
el Flynn wrote: Adnan Ahmed wrote: hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck my configs are. snip *clisip show peers Name/UsernameHost Mask Port Status 101/101192.168.10.195255.255.255.255 5060Unmonitored your sip show peers command shows that the phone is indeed connected to your Asterisk server. If you are having problems doing stuff with it, may I suggest you changing your dialplan to the following just to test things out: [sip] exten = 1,1,VoicemailMain exten = 1,2,Hangup then restart asterisk and dial 1 from your SIP phone. If you can hear the voicemail application prompts then you're okay. flynn *It's not working still in silence mode don't show anything don't do anything any other suggestions,tips ,configs .* Thanks In Advance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to open master device '/dev/zap/ctl'
Jose Hernandez wrote: I installed TDM400P and X100P pci cards in a system running mandrake 10.1 official, kernel 2.6.8.1-12mdksmp. I can compile zaptel, libpri, asterisk and modprobe (zaptel, wcfxs, wcfxo) without errors. Except that running ztcfg and asterisk fails. [EMAIL PROTECTED] asterisk]# ztcfg Notice: Configuration file is /etc/zaptel.conf line 3: Unable to open master device '/dev/zap/ctl' [EMAIL PROTECTED] asterisk]# asterisk -vvvcg ... == Parsing '/etc/asterisk/zapata.conf': Found Nov 22 21:16:11 WARNING[14643]: chan_zap.c:757 zt_open: Unable to open '/dev/zap/channel': No such file or directory Nov 22 21:16:11 ERROR[14643]: chan_zap.c:6195 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp-channel = 1, channel = 1 Nov 22 21:16:11 ERROR[14643]: chan_zap.c:9139 setup_zap: Unable to register channel '1' Nov 22 21:16:11 WARNING[14643]: loader.c:334 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Nov 22 21:16:11 WARNING[14643]: loader.c:429 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] asterisk]# Ouch ... error while writing audio data: : Broken pipe zaptel.conf loadzone = us fxoks = 1 I found an page on the wiki that suggested commenting the ifeq ($(DYNFS),) else end if block in Makefile. I made the changes recompiled without errors but still same error running ztcfg and starting asterisk. Any suggestions? - Jose double check your zaptel.conf file you missing alot like in your TDM400 how many modules you are using 1 fxs/fxo 2 04 more and also in x100p using fxo or fxs after checking these modules properly define in zaptel.conf like fxoks =1 ;x100p fxsks = 3 ;TDM400 etc. hope this helps. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring Asterisk As A Sip Server
Hello Group, I want to configure my Asterisk Server As a SIP is there any possibality.How i do that.Any help is highly appreciated. Thanks in advance. Regards Adnan . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to create channel of type Zap!
hi,
I am using TDM400 with FXS and FXO modules,everytime i try to make call
of my analog phone it gives following errors:
Executing Dial( , ) in
new stack
app_dial.c:554 dial_exec:Unable to create channel of type Zap Everyone
is busy at this time.
Executing Congestion( , ) in new stack
Spawn Extension (outgoing,6943442, 2) exitd non-zero on Zap/1-1 ;in
which 6943442 is local number and '2' i don't know what is that.
Hungup Zap/1-1
I am new in this group and also asterisk too so don't bother with my
questions!
my configs are:
zaptel.conf
fxoks=1
fxsks=4
loadzone=us
defaultzone=us
zapata.conf
[channels]
signalling=fxo_ks
context=outgoing
channel = 1
signalling=fxs_ks
context=incoming
channel = 4
extensions.conf
[incoming]
exten = s,1,Dial,Zap/4
[outgoing]
exten = _NXX,1,Dial/Zap/1/${EXTEN:1}
kindly pointout my mistakes/errors and helping me out.
Thanks In Advance .
Adnan Ahmed.
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[Asterisk-Users] unable to create channel of type Zap
hi,
I am using TDM400 with FXS and FXO modules,everytime i try to make call
of my analog phone it gives following errors:
Executing Dial( , ) in
new stack
app_dial.c:554 dial_exec:Unable to create channel of type Zap Everyone
is busy at this time.
I am new in this group and also asterisk too so don't bother with my
questions!
my configs are:
zaptel.conf
fxoks=1
fxsks=4
loadzone=us
defaultzone=us
zapata.conf
[channels]
signalling=fxo_ks
context=outgoing
channel = 1
signalling=fxs_ks
context=incoming
channel = 4
extensions.conf
[incoming]
exten = s,1,Dial,Zap/1
;exten = s,1,Dial,Zap/4
in above two lines which one os appropriate i am trying both options but
no result.
[outgoing]
exten = 021NXX,1,Dial/Zap/1/${EXTEN:1}
kindly pointout my mistakes/errors and helping me out.
Thanks In Advance .
Adnan Ahmed.
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[Asterisk-Users] Asterisk Hanging!
hi,
everytime i run asterisk and looking asterisk log i found following errors:
parse error: No category context for line 96 of extensions.conf
Requested contexts didn't get merged
Also asterisk not run just initialize and freezes and the log shows
above description.
my configs are:
zaptel.conf
fxoks=1
fxsks=4
loadzone=us
defaultzone=us
zapata.conf
[channels]
signalling=fxo_ks
context=outgoing
channel = 1
signalling=fxs_ks
context=incoming
channel = 4
extensions.conf
[incoming]
exten = s,1,Dial,Zap/1
[outgoing]
exten = _0NXX,1,Dial/1/${EXTEN:1}
I am a newbie so don't be bothering with my configuration.
kindly pointout my mistakes/errors and helping me out.
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[Asterisk-Users] can i call my local phone to IP phone or vice versa
Hello, I am using TDM400 with FXO and FXS modules is there any possibality to call my local PSTN phone to my ip phone or vice versa for what configuration i adopt and if not possible what's the right approach. Thanks in Advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk's Fails to start!
hi,
I am using TDM400 with FXS and FXO modules,everytime i run asterisk and
looking asterisk log i found following errors:
parse error: No category context for line 96 of extensions.conf
Requested contexts didn't get merged
Also asterisk not run just initialize and freezes and the log shows
above description.
my configs are:
zaptel.conf
fxoks=1
fxsks=4
loadzone=us
defaultzone=us
zapata.conf
[channels]
signalling=fxo_ks
context=outgoing
channel = 1
signalling=fxs_ks
context=incoming
channel = 4
extensions.conf
[incoming]
exten = s,1,Dial,Zap/1
[outgoing]
exten = _0NXX,1,Dial/1/${EXTEN:1}
I am a newbie so don't be bothering with my configuration.
kindly pointout my mistakes/errors and helping me out.
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[Asterisk-Users] FXO Module Error
Hi, I am using TDM400 with FXO and FXS ,every time i probe FXO with modprobe wcfxo it gives error that no such device and refers to insmod /dmesg for details but when i dmesg it gives me following description about it. Zapata Telephony Interface Registered on major 196 Freshmaker version: 71 Freshmaker passed register test Module 0:Installed -- Auto FXS/DPO Module 1:Not Installed Module 2:Not Installed Module 3:Installed -- Auto FXO (FCC MODE) Found a Wildcard TDM:Wildcard TDM400 REV E/F (4 modules) Also insmod gives same error PLUS it gives Hint: insmod errors can be caused by incorrect module parameters,including invalid IO or IRQ parameters. What is this meaning ? Hardware or Software problem?I don't know. Kindly helping me out. Thanks in advance. Adnan Ahmed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trunk doesn't work Adit 600/T100P
Hi ! I am connecting to Adit 600 thru a T100P card I have configured 1-16 FXS channels and 17-24 FXO. Everything looks fine on Asterisk side I get a tone on all FXS channels, but when I try to dialout thru one of the FXO channels 17-24 it doesn't connect to the POTS line and echoes back my voice. I use fxsls and fxols for the T1 channels and ls on Adit side. Whats wrong here ? here is my Adit conf voip-pbx print config - -Cactus.lite configuration file -Created on 01/01/1999 at 00:02:49 for adnan -This file is valid for the following configuration only: - -CardType - -SLOT AT1x2 Code Revision: 1.3.1 -SLOT 1FXSx8 -SLOT 2FXSx8 -SLOT 3FXSx8 -SLOT 4FXSx8 -SLOT 5FXOx8 -SLOT 6FXSx8 - -Note: Lines beginning with '-' will be ignored as comme -by the CLI. Before downloading, review the sections of -configuration file delimited by these comments and delet -the commands that are not needed (e.g. 'set ip address' -and 'add user' are likely candidates for deletion). - -While downloading, a character and line delay of 5 ms is -recommended. - -Turning off verification messages. set verification off -Setting local off. set local off -Disconnecting all connections. disconnect a disconnect 1 disconnect 2 disconnect 3 disconnect 4 disconnect 5 disconnect 6 -Setting users. add user adnan -Setting network id. set id voip-pbx -Setting primary and secondary clock sources. set clock1 a:1 set clock2 internal -Setting IP addresses. set ethernet ip address 192.168.7.151 255.255.255.192 set ip gateway 0.0.0.0 -Setting the SNMP MIB-II System Group objects. set snmp getcom public set snmp setcom public set snmp trapcom public -Setting slot a. set a:1 up set a:1 fdl none set a:1 lbo 1 set a:1 framing esf set a:1 id CAC DS1# 01 set a:1 linecode b8zs set a:1 loopdetect on set a:1 threshold min15 uas default set a:1 threshold min15 ses default set a:1 threshold min15 es default set a:1 threshold min15 sefs default set a:1 threshold min15 les default set a:1 threshold min15 css default set a:1 threshold min15 bes default set a:1 threshold min15 dm default set a:1 threshold min15 lcv default set a:1 threshold min15 pcv default set a:1 threshold day uas default set a:1 threshold day ses default set a:1 threshold day es default set a:1 threshold day sefs default set a:1 threshold day les default set a:1 threshold day css default set a:1 threshold day bes default set a:1 threshold day dm default set a:1 threshold day lcv default set a:1 threshold day pcv default set a:1:1-24 signal ls set a:1:1-24 type voice set a:2 down set a:2 fdl none set a:2 lbo 1 set a:2 framing esf set a:2 id CAC DS1# 02 set a:2 linecode b8zs set a:2 loopdetect on set a:2 threshold min15 uas default set a:2 threshold min15 ses default set a:2 threshold min15 es default set a:2 threshold min15 sefs default set a:2 threshold min15 les default set a:2 threshold min15 css default set a:2 threshold min15 bes default set a:2 threshold min15 dm default set a:2 threshold min15 lcv default set a:2 threshold min15 pcv default set a:2 threshold day uas default set a:2 threshold day ses default set a:2 threshold day es default set a:2 threshold day sefs default set a:2 threshold day les default set a:2 threshold day css default set a:2 threshold day bes default set a:2 threshold day dm default set a:2 threshold day lcv default set a:2 threshold day pcv default set a:2:1-24 signal ls set a:2:1-24 type voice -Setting slot 1. set 1:1-8 signal ls set 1:1-8 txgain 0 set 1:1-8 rxgain 0 set 1:1-8 linelength short -Setting slot 2. set 2:1-8 signal ls set 2:1-8 txgain -3 set 2:1-8 rxgain -6 set 2:1-8 linelength short -Setting slot 3. set 3:1-8 signal ls set 3:1-8 txgain -3 set 3:1-8 rxgain -6 set 3:1-8 linelength short -Setting slot 4. set 4:1-8 signal ls set 4:1-8 txgain -3 set 4:1-8 rxgain -6 set 4:1-8 linelength short -Setting slot 5. set 5:1-8 signal ls set 5:1-8 txgain 0 set 5:1-8 rxgain 0 -Setting slot 6. set 6:1-8 signal ls set 6:1-8 txgain -3 set 6:1-8 rxgain -6 set 6:1-8 linelength short -Making connections. connect a:1:1-8 1:1-8 connect a:1:9-16 2:1-8 connect a:1:17-24 5:1-8 -Turning verification on. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Customized Call Parking
Hi ! I need a solution to park incoming calls to an extension of my choice where a special announcement is played, park subsequent calls to specific pools so that they listen to announcements of my choice. any ideas ? Shah. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ALSA help required !
I have just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. --- [chan_alsa.so] = (ALSA Console Channel Driver) Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:474 soundcard_init: Problem opening alsa I/O devices == No sound card detected -- console channel will be unavailable == Turn off ALSA support by adding 'noload=chan_alsa.so' in /etc/asterisk/modules.conf -- earlier when using the OSS, the playback was choppy not smooth, I added some more RAM (total 256 on Intel PIII 600 processor), but the problem was still there so I turned to the Alsa drivers.Asterisk doesn't seem to work with it what might be wrong, any ideas ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Alsa driver doesn't initialize
---BeginMessage--- ---BeginMessage--- I have just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. --- [chan_alsa.so] = (ALSA Console Channel Driver) Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:474 soundcard_init: Problem opening alsa I/O devices == No sound card detected -- console channel will be unavailable == Turn off ALSA support by adding 'noload=chan_alsa.so' in /etc/asterisk/modules.conf -- earlier when using the OSS, the playback was choppy not smooth, I added some more RAM (total 256 on Intel PIII 600 processor), but the problem was still there so I turned to the Alsa drivers.Asterisk doesn't seem to work with it what might be wrong, any ideas ? ---End Message--- ---End Message---
