sr-users October 2005

sr-users@lists.kamailio.org

214 participants
457 discussions

[Users] manipulate $from under avops module

by Brandon Price

I am having some trouble with avpops and the from variable. I am doing something like avp_db_load("$to","$disabled"); which produces a query like select value,attribute,type from usr_preferences where username='+15183207486' AND attribute='50' the problem is I need a query like select value,attribute,type from usr_preferences where username='15183207486' AND attribute='50' ,without the "+" is there any way I can strip the 1st digit from the $from variable? I have already tried using strip(1); As always any help greatly appreciated.

18 years, 8 months

[Serusers] [HELP]:Failed to use X-Pro to call

by shikun.wang

Hi, When I use X-pro to do a call test,the radius server debugs: rad_recv: Access-Request packet from host 127.0.0.1:33407, id=111, length=256 User-Name = "<A href="mailto:003@159.226.58.100">003(a)159.226.58.100</A>" Digest-Attributes = 0x0a05303033 Digest-Attributes = 0x01103135392e3232362e35382e313030 Digest-Attributes = 0x022a34333465343837383763373365376536316365356238636663636135333562643837303161633263 Digest-Attributes = 0x04147369703a3135392e3232362e35382e313030 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303139 Digest-Attributes = 0x08224244303044334532333841363432363341434337414135414643444646454635 Digest-Response = "f4ac6b69ae532a2feb1b8b9e32e0fe86" Service-Type = Sip-Session Sip-Uri-User = "003" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 modcall: entering group authorize modcall[authorize]: module "preprocess" returns ok modcall[authorize]: module "chap" returns noop rlm_eap: EAP-Message not found modcall[authorize]: module "eap" returns noop rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "003" Digest-Realm = "159.226.58.100" Digest-Nonce = "434e48787c73e7e61ce5b8cfcca535bd8701ac2c" Digest-URI = "sip:159.226.58.100" Digest-Method = "REGISTER" Digest-QOP = "auth" Digest-Nonce-Count = "00000019" Digest-CNonce = "BD00D3E238A64263ACC7AA5AFCDFFEF5" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok rlm_realm: Looking up realm "159.226.58.100" for User-Name = "<A href="mailto:003@159.226.58.100">003(a)159.226.58.100</A>" rlm_realm: No such realm "159.226.58.100" modcall[authorize]: module "suffix" returns noop users: Matched DEFAULT at 152 modcall[authorize]: module "files" returns ok modcall[authorize]: module "mschap" returns noop modcall: group authorize returns ok rad_check_password: Found Auth-Type DIGEST auth: type "digest" modcall: entering group authenticate rlm_digest: Configuration item "User-Password" is required for authentication. modcall[authenticate]: module "digest" returns invalid modcall: group authenticate returns invalid auth: Failed to validate the user. Delaying request 34 for 1 seconds Finished request 34 Going to the next request Sending Access-Reject of id 110 to 127.0.0.1:33406 "003" is the user name and I have set the secret in the configure files of both radius server and radius client. Anyone can give me an instruction?Many Thanks.

18 years, 8 months

[Serusers] Smoothly transition from only Asterisk to Openser plus Asterisk

by Ronald Wiplinger

The last week__s I tried to insall webser on a machine other than my asterisk box. I cannot make it!!! My next attempt is to use my Asterisk box for openSER and Asterisk. I will install openser and webser first with listening to port 5062 instead of the usually 5060 SIP port. Than I will test openSER and serweb. So far, I believe I can make it, but for the next steps I need some help. What is the easiest way to get the Asterisk users into SER? I want to swap in Asterisk the port 5060 to 5061 and from openSER the port 5062 to 5060 AND forward all calls to Asterisk. What do I need to do for that? If the user is registered in Asterisk AND in SER, than he will get a different context, while if only in SER, than he should only get [from_ser] which means only local calls. In addition I want to allow for SER users ENUM and DUNDI. I found a module ENUM, but I did not find a DUNDI module. Any ideas, suggestions or recommendations for my path? BTW, I read many docs about SER and it mentions iptel.cfg in the etc directory, but I do not have that example. Where can I get this one? bye Ronald Wiplinger

18 years, 8 months

[Serusers] SRC-Based Routing?

by Duncan Glendinning

I am using SER is the primary SIP server in front of four Asterisk PBX's each located in a different city(e.g., City 1: 899xxxx, City 2: 598xxxx, City 3: 356xxxx, City 4: 829xxxx). Users are identified by a 7 digit number based on their home city. For PSTN-bound destinations, I want to be able to direct SIP calls to the appropriate Asterisk server based on who is calling. Using the src IP address isn't sufficient as users move around the network - in one city one day and another city another day. For example, I want a call from user 8991234(a)domain.com to 91234567(a)domain.com to be sent to the server in City 1 because user 8991234 resides in City 1. I can't use the src ip address because users move around in the network. I have wondered about using a city-based db in addition to a global db. Being a newcomer to this, I'm hoping that there is a simple solution to this problem. Your help and insights would be greatly appreciated, Duncan --- <mailto:Duncan.Glendinning@cox.net> Duncan.Glendinning(a)cox.net

18 years, 8 months

[Serusers] failure route

by Jason Mohler

Is there anyone that can help me with my following syntax? I am basically trying to have all calls come in from an IP and if that IP fails for some reason, it will roll to another IP. if (src_ip== xxx.xxx.xxx.xxx){ t_on_failure("1"); rewritehost("xxx.xxx.xxx.xxx"); }; route[1]{ rewritehost("xxx.xxx.xxx.xxx"); break; t_relay(); };

18 years, 8 months

[Users] server IP and DNS SRV rewrite

by Helge Waastad

Hi, I was just thinking..... I have a multi-domain setup and using domain in lookup(). However, some UAC sends the proxy address in INVITE. If all my contacts are registered with <username> and <domain>, lookup ("location") will return false since the domain in request will be <proxy ip address>..... Is there a way of reverse dns handle this or is just textops the way to go? -- mvh/best regards Helge Waastad System Engineer Smartnet

18 years, 8 months

[Serusers] Missed Call Return

by Aisling

Hello, I read a previous post by Iqbal (missed caller call back) where a subscriber can ring a pre-defined number which calls back their last missed call... Is my logic correct as follows?: In the missed call section (i.e. when theres a 408/487 message) do: avp_db_store("$ruri", "i:1000"); Then when the caller dials the callback number e.g.1471 this code is invoked: if (uri=~"^sip:1471@x.x.x.x"){ log(1, "in missed call return section"); avp_db_load("$from/username", "i:1000"); avp_pushto("i:1000", "$ruri"); // route as normal }; I am getting an error when restarting SER: ERROR: avops: fixup_pushto_avp: bad param 1; expected : $[ruri|hdr_name|..] Is it correct to push the AVP i:1000 to the ruri? Do I need to define aliases in the modparam perhaps? Any help would be appreciated. Many thanks, Aisling. p.s. In case anyone was following my other post regarding asterisk voicemail - It's now working using avpops, the problem was with my asterisk config. There is still a problem playing back the messages though. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

18 years, 8 months

[Serusers] BYE method not handled properly

by Sam Lee

I would like to know why does my BYE method are always replied with a 'Call Leg/Transaction does not exist' . How do they compare whether the transaction in the BYE method exist or not ? ( tag? ftag ? ) Are there any thing in the config that might cause this kind of problem ? Just want to highlight that all the calls are made in a good condition, everything except when the call is ending. I have checked that the phones have not received a prior BYE. Any other idea what is wrong ? Here's a more detailed situation :- Caller (PSTN) --> Voice Gateway --> OPENSER --> Callee (UA) When Callee (UA) tried to end the call , OPENSER will forward a copy of the BYE to Voice Gateway to inform him of the BYE. The Gateway , somehow , replied with a 'Call Leg/Transaction Does Not Exist' . The strange thing is, the Caller (PSTN) was somehow informed of the BYE method and terminate the session . Anyone has any idea how to handle these errors ? I will be glad to provide a ngrep for more reference. Please let me know if you dont understand. Regards, Sam

18 years, 8 months

[Users] Polycom soundpoint ip 500

by Cesc

Hi, We have a polycom sound point ip 500, sip app ver 1.5 (from voip-info.org<http://voip-info.org>). It turns out that it won't display MESSAGE ... but it does receive them and it replies with a 200 OK. But no message on the screen or anywhere within the phone. The configuration has not been changed, it is pretty much the standard stuff that comes from voip-info.org <http://voip-info.org>. Anyone has ever came across the same problem? Regards, Cesc PS - Getting the sip software from polycom resellers is a mess ... well, i would say almost impossible. Next time we'll go for a costumer frendlier phone manufacturer i guess.

18 years, 8 months

Re: [Serusers] Have some INVITE problems

by Samuel Osorio Calvo

I'm sorry but I have never played with these models so I can not help you much..... In order to extend the INVITE retransmission, there must be somewhere in the web interface of these phones (if they have) something like T1 and T2, these values determine the retransmission period. I guess you should check the Internet for information about your phones. Hope somebody else with more experience can help you, Samuel. Unclassified. >>> Alex <alexandergav(a)gmail.com> 10/11/05 05:45PM >>> I am using cisco 7960 and grandstream. 286,486 On 10/11/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > > Which UA are you using??? It looks like all your issues comes from the > implementation you are using as end-point and has nothing to do with SER. > > Samuel. > > > Unclassified. > >>> Alex <alexandergav(a)gmail.com> 10/11/05 11:06AM >>> > Hi all > I would like to explain some problem. > When my UAC initiates INVITE the server response with 100 trying > After receiving 100 trying the UAC sending again INVITE. ( HERE is the > problem). > RFC - 3261 ("After receiving a 1xx response, any retransmissions cease > altogether "); > Any help on that will be appreciated. > In addition if there any way to extend the timeout between the INVITES. > Thanks > >

18 years, 8 months

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sr-users October 2005