similar to: SIP endpoint registeration problem

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

Hi List; I noticed that if I disabled secret in the context by placing ( ; ) before it, then at the asterisk the log will be: -- Registered SIP 'bilal_sip' at 0.0.0.0 port 5060 expired The IP address of the endpoint was not captured!!! Why? If secret enabled, then some endpoints can not register (maybe due to compatibility in reading the negotiation packets), so what is the solution?

Dear Philipp; Kindly find the part of the configuration as below: [general] allow=all disallow is comment by ( ; ). [bilal_sip] type=friend context=internal host=dynamic canreinvite=no dtmfmode=rfc2833 So where is the problem? The endpoint does not register and nothing appear on trace level 3. And the amazing thing that if the endpoint send wrong username (for example: bilal_sip100) then it

help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal

Hi List; How can I configure asterisk to receive a call from SIP end point without being registered at asterisk and its IP address is dynamic, and authentication to be based on the username and password or any other string? I know that if I place the host with static IP then no need to register, but what if the voip gateway was having dynamic IP and I do not need to register on asterisk, but I

Dear Alex; Thanks alot for your nice help. This is if I need to let Asterisk register with another softswitch (so I used register =>), what if I need asterisk to send call for the softswitch without register to it (directly)? If I removed the register => then how it will distiguish the IP address in the "host" at the [sip_trunk] is the IP address of the softswitch that need to

Dear List; Thanks alot for the help. But how can I let the second dial tone (after pressing the extension to select that FXO port) to be difference than normal dial tone? Regards Bilal Ghayad -------------------------- Correction, on FXO port not FXS, second, read his email first: "Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

Dear List; How can I place a call via Zap/g1 (group) but need to determine the line (FXO port) that will go via it? Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP Phone or Broadtel IP Phone, so if user select that button then he will be sure that his outside call will be via that specific line. Regards Bilal

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040108/748d21b3/attachment.htm -------------- next part -------------- Hello I downloaded latest CVS version and got following error messages while registeration. ( CVS Version Asterisk CVS-01/08/04-14:20:34) As a result IP Phone don't register with the Asterisk. Is it broken ? How can I

Hi List; Where I determine the codec to be used for the SIP Trunk (between Asterik and another SIP softswitch)? Regards Bilal ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

I use the E-series Nokia phones on my Wireless LAN. The e series have sip agent On 8/20/07, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Dears; To patch libpri: I just place the patch file in the libpri source directory and then I run make and make install? Or I need to compile the dahdi and asterisk also? If the problem stayed, do I have to go for previous libpri version? Or for previous dahdi version and asterisk version? Regards Bilal ----------- > bilal ghayyad wrote: > > But I am afraid it is a bug because I

Hi Danny; Really I did not understand how I can determine if the IO will be DB or conf files? Is it from the Asterisk manager? Regards Bilal ------------------------- It depends on how you are configured. The gui interfaces using Asterisk Manager, so you get the Same IO from the gui that you would get from a native manager session. -----Original Message----- From: asterisk-users-bounces at

Hi All; Alot of softswitches or PBX's does not accept to manipulate any SIP call without being registered firstly. So that means, I need asterisk to register firstly then I can route my calls to that SIP trunk. In IAX2, we use the register => , so what shall we do in Asterisk? And how its format will be (if we will use register)? Or what is the solution? Regards Bilal